RE: I new two pipe line here for rtp sending and receiving , but meet some problems
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-----
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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]"
<[hidden email]>
Subject: [gst-devel] Problem Trying To Use v4l2src
To: "[hidden email]"
<[hidden email]>
Message-ID:
<[hidden email]>
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 <[hidden email]>
Subject: Re: [gst-devel] Problem Trying To Use v4l2src
To: [hidden email]
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" <[hidden email]>
Subject: Re: [gst-devel] Build ffmpeg plugin with x264 static lib
To: Discussion of the development of GStreamer
<[hidden email]>
Message-ID:
<[hidden email]>
Content-Type: text/plain; charset="gb2312"
Hi
can the ugly plugin work well?
-----Original Message-----
From: Edward Hervey [[hidden email]]
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
>
>
>
>
> ------------------------------------------------------------------------------
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------------------------------
Message: 4
Date: Wed, 19 Aug 2009 10:11:03 +0800
From: "Zhang, Boning" <[hidden email]>
Subject: [gst-devel] I new two pipe line here for rtp sending and
receiving , but meet some problems
To: "[hidden email]"
<[hidden email]>
Message-ID:
<[hidden email]>
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 http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer.
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
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Message: 5
Date: Tue, 18 Aug 2009 22:09:22 -0500 (CDT)
From: Nathan Stratton <[hidden email]>
Subject: [gst-devel] BlackMagic Intensity
To: [hidden email]
Message-ID: <[hidden email]>
Content-Type: TEXT/PLAIN; format=flowed; charset=US-ASCII
Looking for someone to write plugin for the BlackMagic Intensity Pro card:
http://www.blackmagic-design.com/products/intensity/
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
http://www.robotics.net http://www.blinkmind.com
------------------------------
Message: 6
Date: Tue, 18 Aug 2009 23:06:03 -0500
From: "Gao, Ping" <[hidden email]>
Subject: [gst-devel] rtspsrc doesn't show in gst-inspect on target
system
To: "[hidden email]"
<[hidden email]>
Message-ID:
<[hidden email]>
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 <[hidden email]>
Subject: [gst-devel] no audio when file with audio-video streams is
played
To: Discussion of the development of GStreamer
<[hidden email]>
Message-ID:
<[hidden email]>
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
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