<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN"> <HTML> <HEAD> <META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=iso-8859-1"> <META NAME="Generator" CONTENT="MS Exchange Server version 6.5.7653.38"> <TITLE>RE: I new two pipe line here for rtp sending and receiving , but meet some problems</TITLE> </HEAD> <BODY>   Hello Zhang,  From the log you have mentioned , it seems you probably need the caps  to be set in between udpsrc and depay in the receiver side  You may try the following and see if it works if you have not done it already -  Receving:  gst-launch udpsrc port=5000 ! application/x-rtp, media=video, payload=X, clock-rate=90000, encoding-name=H263 ! rtph263depay ! ffdec_h263 ! filesink location=./abc.out  Where X = 34/96/127 . Please check and replace accordingly.  Regards,  Arnab -----Original Message----- From: gstreamer-devel-request@lists.sourceforge.net [<A HREF="mailto:gstreamer-devel-request@lists.sourceforge.net">mailto:gstreamer-devel-request@lists.sourceforge.net</A>] Sent: Wed 8/19/2009 11:32 AM To: gstreamer-devel@lists.sourceforge.net Subject: gstreamer-devel Digest, Vol 39, Issue 23 Send gstreamer-devel mailing list submissions to         gstreamer-devel@lists.sourceforge.net application/x-rtp                   media: video                 payload: 34              clock-rate: 90000           encoding-name: H263 To subscribe or unsubscribe via the World Wide Web, visit         <A HREF="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</A> or, via email, send a message with subject or body 'help' to         gstreamer-devel-request@lists.sourceforge.net You can reach the person managing the list at         gstreamer-devel-owner@lists.sourceforge.net When replying, please edit your Subject line so it is more specific than "Re: Contents of gstreamer-devel digest..." Today's Topics:    1. Problem Trying To Use v4l2src       (Kulecz, Walter (JSC-SK)[WYLE LABORATORIES])    2. Re: Problem Trying To Use v4l2src (Tim-Philipp M?ller)    3. Re: Build ffmpeg plugin with x264 static lib (Zhang, Boning)    4. I new two pipe line here for rtp sending and receiving , but       meet some problems (Zhang, Boning)    5. BlackMagic Intensity (Nathan Stratton)    6. rtspsrc doesn't show in gst-inspect on target system (Gao, Ping)    7. no audio when file with audio-video streams is played (Jyoti) ---------------------------------------------------------------------- Message: 1 Date: Tue, 18 Aug 2009 17:30:16 -0500 From: "Kulecz, Walter (JSC-SK)[WYLE LABORATORIES]"         <walter.kulecz-1@nasa.gov> Subject: [gst-devel] Problem Trying To Use v4l2src To: "gstreamer-devel@lists.sourceforge.net"         <gstreamer-devel@lists.sourceforge.net> Message-ID:         <1F2262C21D3BB74DBE5C9576EAB4BF4F944D3AC7A5@NDJSSCC03.ndc.nasa.gov> Content-Type: text/plain; charset="us-ascii" I'm evaluating the possibility of using gstreamer for the next version of my real time image processing application.  It seems I need to write a gstreamer plugin to get access to the actual data being passed through the gstreamer pipeline. I've succeed in using the gst-template to compile a simple filter plugin, but before I put much more effort into learning gstreamer I need to solve a problem with the v4l2src plugin. My question is how do I initialize the plugin?  I need to set NTSC mode (some of our cards default to PAL on powerup), image size to 640x480 (although it seems I could use one of the resize or cropping plugins),  select the composite or S-Video input (most cards seem to default to tuner), and maybe set 8-bit grayscale video mode (our cameras are monochrome).  If I run an application like TVtime first, the card then works with v4l2src in a simple gst-launch pipeline to test my plugin.  But it seems pretty fragile with respect to video errors often requiring another run of TVtime to reset the card so v4l2src will work. I've searched the gestreamer-devel archive and only found some discussion of a similar issue on Win32 with directshow tuners. ------------------------------ Message: 2 Date: Wed, 19 Aug 2009 00:13:45 +0100 From: Tim-Philipp M?ller <t.i.m@zen.co.uk> Subject: Re: [gst-devel] Problem Trying To Use v4l2src To: gstreamer-devel@lists.sourceforge.net Message-ID: <1250637225.19902.26.camel@zingle> Content-Type: text/plain On Tue, 2009-08-18 at 17:30 -0500, Kulecz, Walter wrote: > I'm evaluating the possibility of using gstreamer for the next version > of my real time image processing application.  It seems I need to > write a gstreamer plugin to get access to the actual data being passed > through the gstreamer pipeline. Not necessarily, but if you want to manipulate the data in the middle of a pipeline writing an element is usually the right thing to do. You can get data out of a pipeline using the appsink element, or the fakesink element with a "handoff" signal handler, or the identity element with a "handoff" signal handler, or by setting up a pad probe. > I've succeed in using the gst-template to compile a simple filter > plugin, but before I put much more effort into learning gstreamer I > need to solve a problem with the v4l2src plugin. > > My question is how do I initialize the plugin?  I need to set NTSC mode > (some of our cards default to PAL on powerup), image size to 640x480 > (although it seems I could use one of the resize or cropping > plugins),  select the composite or S-Video input (most cards seem to > default to tuner), and maybe set 8-bit grayscale video mode (our > cameras are monochrome).  The desired pixel format/layout and resolution can be set by putting a capsfilter with suitable filter caps after the v4l2src element. This usually assumes the camera/drivers/libv4l support the exact format and resolution desired. If you just want to get data in a particular format/size no matter what you can put some converters like ffmpegcolorspace and videoscale between v4l2src and your capsfilter, and those elements will try their best to convert the data from the camera into the desired format if the format isn't supported. NTSC/PAL selection is done using the GstTuner interface (see gst-plugins-base libraries docs), I believe. The device might need to be open already when you use the interface (ie. the v4l2src element needs to be set to at least READY or PAUSED state [it should be READY, but IIRC the v4l elemens didn't implement that correctly until recently, so you might need PAUSED instead depending on the version used]). > If I run an application like TVtime first, the card then works with > v4l2src in a simple gst-launch pipeline to test my plugin.  But it > seems pretty fragile with respect to video errors often requiring > another run of TVtime to reset the card so v4l2src will work. What errors do you get? What does TVtime do to reset it? Cheers  -Tim ------------------------------ Message: 3 Date: Wed, 19 Aug 2009 09:44:16 +0800 From: "Zhang, Boning" <boning.zhang@intel.com> Subject: Re: [gst-devel] Build ffmpeg plugin with x264 static lib To: Discussion of the development of GStreamer         <gstreamer-devel@lists.sourceforge.net> Message-ID:         <037F493892196B458CD3E193E8EBAD4F01EC3AACFD@pdsmsx502.ccr.corp.intel.com>         Content-Type: text/plain; charset="gb2312" Hi can the ugly plugin  work well? -----Original Message----- From: Edward Hervey [<A HREF="mailto:bilboed@gmail.com">mailto:bilboed@gmail.com</A>] Sent: 2009?8?18? 1:27 To: Discussion of the development of GStreamer Subject: Re: [gst-devel] Build ffmpeg plugin with x264 static lib Hi,   ... don't do that. Just use the x264enc plugin from gst-plugins-ugly.     Edward On Mon, 2009-08-17 at 09:42 +0800, Zhang, Boning wrote: > I try to build the x264 in the ffmpeg plugin,and I change the > configure file in the file with --enable-libx264 > And I can build it and make it to rpm.But after I install it in the > other computer with it rpm, I run gst-inspect : >  > GStreamer-WARNING **: Failed to load plugin > '/usr/lib/gstreamer-0.10/libgstffmpeg.so':/usr/lib/gstreamer-0.10/libgstffmpeg.so:undefined symbol :x264_encoder_encode >  > If I install the ffmpeg-plugin that is OK without any error. > So can any one help me? >  > Thanks > ------------------------------------ > Boning,Zhang >  >  >  >  > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now.  <A HREF="http://p.sf.net/sfu/bobj-july">http://p.sf.net/sfu/bobj-july</A> > _______________________________________________ gstreamer-devel mailing list gstreamer-devel@lists.sourceforge.net <A HREF="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</A> ------------------------------------------------------------------------------ Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day trial. Simplify your report design, integration and deployment - and focus on what you do best, core application coding. Discover what's new with Crystal Reports now.  <A HREF="http://p.sf.net/sfu/bobj-july">http://p.sf.net/sfu/bobj-july</A> _______________________________________________ gstreamer-devel mailing list gstreamer-devel@lists.sourceforge.net <A HREF="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</A> ------------------------------ Message: 4 Date: Wed, 19 Aug 2009 10:11:03 +0800 From: "Zhang, Boning" <boning.zhang@intel.com> Subject: [gst-devel] I new two pipe line here for rtp sending and         receiving , but meet some problems To: "gstreamer-devel@lists.sourceforge.net"         <gstreamer-devel@lists.sourceforge.net> Message-ID:         <037F493892196B458CD3E193E8EBAD4F01EC3AAD34@pdsmsx502.ccr.corp.intel.com>         Content-Type: text/plain; charset="us-ascii" sending: gst-launch videotestsrc ! videoscale ! video/x-raw-yuv,width=352,height=288 ! videorate ! video/x-raw-yuv,framerate=15/1 ! videobalance brightness=0.3 contrast=1.5 ! ffmpegcolorspace ! ffenc_h263! rtph263pay ! udpsink host=localhost port=5000 receving: gst-launch udpsrc port=5000 ! rtph263depay ! ffdec_h263 ! filesink location=./abc.out Problems: They can run separately well, though no data. But If I run them together , it will crash. log: New clock: GstSystemClock ERROR: from element /GstPipeline:pipeline0/GstRtpH263Depay:rtph263depay0: Internal GStreamer error: negotiation problem.  Please file a bug at <A HREF="http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer">http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer</A>. Additional debug info: gstbasertpdepayload.c(360): gst_base_rtp_depayload_chain (): /GstPipeline:pipeline0/GstRtpH263Depay:rtph263depay0: Not RTP format was negotiated Execution ended after 37364529 ns. Can anyone tell me how to solve this problem? Thanks very much. ------------------------------------ Boning,Zhang -------------- next part -------------- An HTML attachment was scrubbed... ------------------------------ Message: 5 Date: Tue, 18 Aug 2009 22:09:22 -0500 (CDT) From: Nathan Stratton <nathan@robotics.net> Subject: [gst-devel] BlackMagic Intensity To: gstreamer-devel@lists.sourceforge.net Message-ID: <alpine.LFD.2.00.0908182206460.3811@bart.robotics.net> Content-Type: TEXT/PLAIN; format=flowed; charset=US-ASCII Looking for someone to write plugin for the BlackMagic Intensity Pro card: <A HREF="http://www.blackmagic-design.com/products/intensity/">http://www.blackmagic-design.com/products/intensity/</A> They have a Linux SDK, If anyone is interested, please let me know the est cost. ><> Nathan Stratton                                CTO, BlinkMind, Inc. nathan at robotics.net                         nathan at blinkmind.com <A HREF="http://www.robotics.net">http://www.robotics.net</A>                        <A HREF="http://www.blinkmind.com">http://www.blinkmind.com</A> ------------------------------ Message: 6 Date: Tue, 18 Aug 2009 23:06:03 -0500 From: "Gao, Ping" <ping.gao@harman.com> Subject: [gst-devel] rtspsrc doesn't show in gst-inspect on target         system To: "gstreamer-devel@lists.sourceforge.net"         <gstreamer-devel@lists.sourceforge.net> Message-ID:         <6DF262F0BAB5944AB03D1FE91354A1EF05FF1AEAD0@HICGWSEX01.ad.harman.com> Content-Type: text/plain; charset="Windows-1252" Hi there: I am trying to add RTSP support to a target platform which is x86 based hardware and runs Linux 2.6.28.   The SDK for the target come with g-streamer 0.10 with some elements, but without rtspsrc, however, the libraries are built and exist, just not included in the root fs on the target.    So I copied all pre-built rtspsrc libraries (libgstrtsp-0.10.so*) to where GST_PLUGIN_PATH environment variable point to.   Also, I checked the dependencies of libgstrtsp-0.10.so. The only library that the target missing is libselinux.so.1.  The SDK for the target doesn?t have libselinux.so.1, then I just copied the one on my Linux host (which runs Fedora 9) to my target. When I run ?gst-inspect rtspsrc? on the target, it says ?No such element or plugin ?rtspsrc??.  Can you tell why I don?t have rtspsrc on the target? Any suggestions on how to fix the problem? Thanks a lot. Ping ------------------------------ Message: 7 Date: Wed, 19 Aug 2009 11:09:39 +0530 From: Jyoti <jyoti.d@allaboutif.com> Subject: [gst-devel] no audio when file with audio-video streams is         played To: Discussion of the development of GStreamer         <gstreamer-devel@lists.sourceforge.net> Message-ID:         <d10c7b710908182239s4ead2c13ta7476e2de5e36270@mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Hi all, I am writing an application to play a media file. 1. The audio doesn't play at all when both audio & video pads generated by demuxer are connected to separate bins called audio & video bins. 2. when I discard the video pad and connect only audio pad to audio bin I am able to hear the audio. The same file plays well with playbin. And I am using the elements as in playbin. *Observation:* 1. On catching bus messages for warning I get this message "*Compensating for audio synchronisation problems*" 2. The video pad is generated first & then the audio pad *The playbin elements are as below:* filesrc->mpegtsdemux ->ffdec_h264->ffmpegcolorspace->videoscale->xvimagesink ->mad->audioconvert->audioresample->alsasink Can someone please suggest some ideas on this? Thanks, Jyoti -------------- next part -------------- An HTML attachment was scrubbed... ------------------------------ ------------------------------------------------------------------------------ Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day trial. Simplify your report design, integration and deployment - and focus on what you do best, core application coding. Discover what's new with Crystal Reports now.  <A HREF="http://p.sf.net/sfu/bobj-july">http://p.sf.net/sfu/bobj-july</A> ------------------------------ _______________________________________________ gstreamer-devel mailing list gstreamer-devel@lists.sourceforge.net <A HREF="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</A> End of gstreamer-devel Digest, Vol 39, Issue 23 *********************************************** <pre> </pre></BODY> </HTML>