Hi, I got the video working well in the following gst-lauch!!! gst-launch -v gnomevfssrc location="<a href="http://carinov:carinov123@10.0.0.150:8000/nphMotionJpeg?Resolution=320x240&Quality=Standard">http://carinov:carinov123@10.0.0.150:8000/nphMotionJpeg?Resolution=320x240&Quality=Standard</a>" do-timestamp=true ! multipartdemux ! image/jpeg,width=240,height=320,framerate=30/1 ! jpegdec ! videorate ! video/x-raw-yuv,framerate=30/1 ! ffenc_mpeg4 ! avimux ! filesink location=a.avi But when i transferd it to code ,i am getting the following error message <unknown>:30333): GStreamer-CRITICAL **: gst_structure_empty_new: assertion `gst_structure_validate_name (name)' failed i am getting this error,,but i didnt used these two function in my program! i am using two capsfilters for setting ! image/jpeg,width=240,height=320,framerate=30/1 ! and setting ! video/x-raw-yuv,framerate=30/1 ! tdst_Vpipeline->flt=gst_element_factory_make("capsfilter","imagefilter"); g_object_set(G_OBJECT(tdst_Vpipeline->flt),"caps",gst_caps_new_simple("image/jpeg", "width",G_TYPE_INT,tdst_Vpipeline->iWidth, "height",G_TYPE_INT,tdst_Vpipeline->iHeight, "framerate",GST_TYPE_FRACTION,tdst_Vpipeline->iFrame_rate,1, NULL),NULL); tdst_Vpipeline->flt2=gst_element_factory_make("capsfilter","videofilter"); g_object_set(G_OBJECT(tdst_Vpipeline->flt),"caps",gst_caps_new_simple("video/x-raw-yuv", "width",G_TYPE_INT,tdst_Vpipeline->iWidth, "height",G_TYPE_INT,tdst_Vpipeline->iHeight, "framerate",GST_TYPE_FRACTION,tdst_Vpipeline->iFrame_rate,1, "mpegversion",G_TYPE_INT,4, NULL),NULL); <div class="gmail_quote">On Mon, May 4, 2009 at 4:30 PM, <<a href="mailto:gstreamer-devel-request@lists.sourceforge.net">gstreamer-devel-request@lists.sourceforge.net</a>> wrote: <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Send gstreamer-devel mailing list submissions to <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> To subscribe or unsubscribe via the World Wide Web, visit <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a> or, via email, send a message with subject or body 'help' to <a href="mailto:gstreamer-devel-request@lists.sourceforge.net">gstreamer-devel-request@lists.sourceforge.net</a> You can reach the person managing the list at <a href="mailto:gstreamer-devel-owner@lists.sourceforge.net">gstreamer-devel-owner@lists.sourceforge.net</a> When replying, please edit your Subject line so it is more specific than "Re: Contents of gstreamer-devel digest..." Today's Topics: 1. What happens when set always_copy=FALSE in v4l2src (yangsb) 2. What happens when set always_copy=FALSE in v4l2src (yangsb) 3. adder with nonblocking pads (Viraj Karandikar) 4. Re: adder with nonblocking pads (Edward Hervey) 5. Re: Sync spectrum messages with audio being played? (Sebastian Dr?ge) 6. Re: How to save a stream buffer to a file ? (Nicolas Bertrand) 7. GTK GUI+ gstreamer (Vikas Patel) ---------------------------------------------------------------------- Message: 1 Date: Sun, 3 May 2009 18:47:35 -0700 (PDT) From: yangsb <<a href="mailto:yangsb05@gmail.com">yangsb05@gmail.com</a>> Subject: [gst-devel] What happens when set always_copy=FALSE in v4l2src To: <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> Message-ID: <<a href="mailto:23361935.post@talk.nabble.com">23361935.post@talk.nabble.com</a>> Content-Type: text/plain; charset=us-ascii Hi. I am testing rtsp-server for streaming out live video on my arm board. The server works fine when always_copy is TRUE. Several days ago, I set always_copy to FALSE , I found that the rtsp server run into a dead state. I tried to debug , and I found that the kernal can not return after executing one videobuf_waiton( ) function . ############################# if (list_empty(&q->stream)) goto done; buf = list_entry(q->stream.next, struct videobuf_buffer, stream); retval = videobuf_waiton(buf, nonblocking, 1); if (retval < 0) ########################### this line of code is in function videobuf_dqbuf() . And the function is in drivers/media/video/video-buf.c However when I use filesink instead of rtsp streaming , the program works fine. My gst-launch is the following : gst-launch \ v4l2src always_copy=FALSE ! \ TIVidenc codecName=h264enc genTimeStamps=1 engineName=encode \ contiguousInputFrame=TRUE iColorSpace=UYVY resolution=640x480 ! \ filesink location=test.264 My rtsp-server is from <a href="http://blogs.gnome.org/uraeus/2008/10/13/gstreamer-rtsp-server/" target="_blank">http://blogs.gnome.org/uraeus/2008/10/13/gstreamer-rtsp-server/</a> Hope to get some advice .. -- View this message in context: <a href="http://www.nabble.com/What-happens-when-set-always_copy%3DFALSE-in-v4l2src-tp23361935p23361935.html" target="_blank">http://www.nabble.com/What-happens-when-set-always_copy%3DFALSE-in-v4l2src-tp23361935p23361935.html</a> Sent from the GStreamer-devel mailing list archive at Nabble.com. ------------------------------ Message: 2 Date: Sun, 3 May 2009 18:46:19 -0700 (PDT) From: yangsb <<a href="mailto:yangsb05@gmail.com">yangsb05@gmail.com</a>> Subject: [gst-devel] What happens when set always_copy=FALSE in v4l2src To: <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> Message-ID: <<a href="mailto:23361935.post@talk.nabble.com">23361935.post@talk.nabble.com</a>> Content-Type: text/plain; charset=us-ascii Hi. I am testing rtsp-server for streaming out live video on my arm board. The server works fine when always_copy is TRUE. Several days ago, I set always_copy to FALSE , I found that the rtsp server run into a dead state. I tried to debug , and I found that the kernal can not return after executing one videobuf_waiton( ) function . ############################# if (list_empty(&q->stream)) goto done; buf = list_entry(q->stream.next, struct videobuf_buffer, stream); retval = videobuf_waiton(buf, nonblocking, 1);retval = videobuf_waiton(buf, nonblocking, 1); if (retval < 0) ########################### this line of code is in function videobuf_dqbuf() . And the function is in drivers/media/video/video-buf.c However when I use filesink instead of rtsp streaming , the program works fine. My gst-launch is the following : gst-launch \ v4l2src always_copy=FALSE ! \ TIVidenc codecName=h264enc genTimeStamps=1 engineName=encode \ contiguousInputFrame=TRUE iColorSpace=UYVY resolution=640x480 ! \ filesink location=test.264 My rtsp-server is from <a href="http://blogs.gnome.org/uraeus/2008/10/13/gstreamer-rtsp-server/" target="_blank">http://blogs.gnome.org/uraeus/2008/10/13/gstreamer-rtsp-server/</a> Hope to get some advice .. -- View this message in context: <a href="http://www.nabble.com/What-happens-when-set-always_copy%3DFALSE-in-v4l2src-tp23361935p23361935.html" target="_blank">http://www.nabble.com/What-happens-when-set-always_copy%3DFALSE-in-v4l2src-tp23361935p23361935.html</a> Sent from the GStreamer-devel mailing list archive at Nabble.com. ------------------------------ Message: 3 Date: Mon, 4 May 2009 10:13:23 +0530 From: Viraj Karandikar <<a href="mailto:virajk@gmail.com">virajk@gmail.com</a>> Subject: [gst-devel] adder with nonblocking pads To: <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> Message-ID: <<a href="mailto:ecfff7140905032143g41c9a697t3a50b2effe1dc894@mail.gmail.com">ecfff7140905032143g41c9a697t3a50b2effe1dc894@mail.gmail.com</a>> Content-Type: text/plain; charset="iso-8859-1" Hi, I want to use adder element in my system. But the adder element waits for buffers on all of its sink pads (for eg: sink0 sink1) , and only when buffers are available on all pads it goes ahead with processing and pushes a output buffer on source pad. What i would like to do is: wait till buffer is availabe on sink0, read buffer from sink1 only if available and do processing. that means, dont wait for any buffer on sink1. I thought of using pull mode on sink1. but i could not find any function to query if buffer is available on a pad. Does anyone has any clue how this can be done? -- - Viraj Reality is merely an illusion, albeit a very persistent one. -------------- next part -------------- An HTML attachment was scrubbed... ------------------------------ Message: 4 Date: Mon, 04 May 2009 08:25:55 +0200 From: Edward Hervey <<a href="mailto:bilboed@gmail.com">bilboed@gmail.com</a>> Subject: Re: [gst-devel] adder with nonblocking pads To: Discussion of the development of GStreamer <<a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a>> Message-ID: <1241418356.722.2.camel@joder> Content-Type: text/plain Hi, You might want to have a look at liveadder (in gst-plugins-bad), which allows doing mixing of live streams. Instead of waiting for data to be available on all streams, it waits up to a certain (configurable) delay before processing data on any incoming streams. Edward On Mon, 2009-05-04 at 10:13 +0530, Viraj Karandikar wrote: > Hi, > I want to use adder element in my system. But the adder element waits > for buffers on all of its sink pads (for eg: sink0 sink1) , and only > when buffers are available on all pads it goes ahead with processing > and pushes a output buffer on source pad. > What i would like to do is: wait till buffer is availabe on sink0, > read buffer from sink1 only if available and do processing. that > means, dont wait for any buffer on sink1. > I thought of using pull mode on sink1. but i could not find any > function to query if buffer is available on a pad. > Does anyone has any clue how this can be done? > > > -- > - Viraj > Reality is merely an illusion, albeit a very persistent one. > ------------------------------------------------------------------------------ > Register Now & Save for Velocity, the Web Performance & Operations > Conference from O'Reilly Media. Velocity features a full day of > expert-led, hands-on workshops and two days of sessions from industry > leaders in dedicated Performance & Operations tracks. Use code vel09scf > and Save an extra 15% before 5/3. <a href="http://p.sf.net/sfu/velocityconf" target="_blank">http://p.sf.net/sfu/velocityconf</a> > _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a> ------------------------------ Message: 5 Date: Mon, 04 May 2009 10:04:06 +0200 From: Sebastian Dr?ge <<a href="mailto:sebastian.droege@collabora.co.uk">sebastian.droege@collabora.co.uk</a>> Subject: Re: [gst-devel] Sync spectrum messages with audio being played? To: Discussion of the development of GStreamer <<a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a>> Message-ID: <1241424246.18835.59.camel@odin.lan> Content-Type: text/plain; charset="us-ascii" Am Sonntag, den 03.05.2009, 12:58 -0700 schrieb EmbeddedMicro: > What I really need is to be able to adjust the time it send the data relative > to when the audio will be played. For example it will send it 100ns before, > or maybe 250ns before, allowing time for the data to be transfered so it is > synced properly. > > It does not seem like this should be too hard to do, but I can't find any > documentation on how to do it in python. Do I have to compare the > running-time with the actual clock or some how set it up so it is called at > even intervals? Take a look at gst-plugins-good/tests/examples/spectrum. The audiotestsrc/osssrc samples are syncing the messages to the clock by installing callbacks into the clock of the pipeline (search for wait_async ) -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: Dies ist ein digital signierter Nachrichtenteil ------------------------------ Message: 6 Date: Mon, 04 May 2009 12:00:25 +0200 From: Nicolas Bertrand <<a href="mailto:nico@inattendu.org">nico@inattendu.org</a>> Subject: Re: [gst-devel] How to save a stream buffer to a file ? To: Discussion of the development of GStreamer <<a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a>> Message-ID: <<a href="mailto:49FEBCB9.5070708@inattendu.org">49FEBCB9.5070708@inattendu.org</a>> Content-Type: text/plain; charset=ISO-8859-1 > Hi Find attached a standalone python program. > This program display a stream ( videotestsec) and allow also making a > snapshot of th stream ( Capture button) . the image result is in > snapshot.jpg > > > The snapshot is made via method capture in class my_gst. Is that the > correct way to that kind of operation ; i.e. storibng an image from > stream in file. > > Actually my question is to know if for save to file a single gstreamer buffer, in an asynchronous way, is better to made this with the add_buffer_probe/remove_buffer_probe methods or use a connect/disconnect 'handoff" signal ? Is there differences ? a more datasafe mode ? gstreamerly mode ? Nico. ------------------------------ Message: 7 Date: Mon, 4 May 2009 16:14:53 +0530 From: "Vikas Patel" <<a href="mailto:Vikas.Patel@imgtec.com">Vikas.Patel@imgtec.com</a>> Subject: [gst-devel] GTK GUI+ gstreamer To: "Discussion of the development of GStreamer" <<a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a>> Message-ID: <<a href="mailto:C1A102308DC77D40BA4DFB4CF4C63BB2567726@pumail1.pu.imgtec.org">C1A102308DC77D40BA4DFB4CF4C63BB2567726@pumail1.pu.imgtec.org</a>> Content-Type: text/plain; charset="us-ascii" Hi, I developed a GUI on GTK+ framework for playing the MPEG2 compressed file on top of Gstreamer plugin That was working fine some time before but now it's giving the segmentation fault in the pipeline. But in this time span I have updated the system with many new release. So does any body have any clue? Regards Vikas - This message is subject to Imagination Technologies' e-mail terms: <a href="http://www.imgtec.com/e-mail.htm" target="_blank">http://www.imgtec.com/e-mail.htm</a> Imagination Technologies Ltd is a limited company registered in England No: 1306335 Registered Office: Imagination House, Home Park Estate, Kings Langley, Hertfordshire, WD4 8LZ. Email to and from the company may be monitored for compliance and other administrative purposes. - -------------- next part -------------- An HTML attachment was scrubbed... ------------------------------ ------------------------------------------------------------------------------ Register Now & Save for Velocity, the Web Performance & Operations Conference from O'Reilly Media. Velocity features a full day of expert-led, hands-on workshops and two days of sessions from industry leaders in dedicated Performance & Operations tracks. Use code vel09scf and Save an extra 15% before 5/3. <a href="http://p.sf.net/sfu/velocityconf" target="_blank">http://p.sf.net/sfu/velocityconf</a> ------------------------------ _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a> End of gstreamer-devel Digest, Vol 36, Issue 4 ********************************************** </blockquote></div> -- Sreerenj B <a href="http://sreerenj.livejournal.com">http://sreerenj.livejournal.com</a> <a href="mailto:bsreerenj@gmail.com">bsreerenj@gmail.com</a> mob: +91 9995377714