[gst-devel] mjpeg streaming error in videorate
sreerenj b
bsreerenj at gmail.com
Mon May 4 15:39:01 CEST 2009
Hi,
I got the video working well in the following gst-lauch!!!
gst-launch -v gnomevfssrc location="
http://carinov:carinov123@10.0.0.150:8000/nphMotionJpeg?Resolution=320x240&Quality=Standard"
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);
On Mon, May 4, 2009 at 4:30 PM, <
gstreamer-devel-request at lists.sourceforge.net> wrote:
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> 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 <yangsb05 at gmail.com>
> Subject: [gst-devel] What happens when set always_copy=FALSE in
> v4l2src
> To: gstreamer-devel at lists.sourceforge.net
> Message-ID: <23361935.post at talk.nabble.com>
> 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
> http://blogs.gnome.org/uraeus/2008/10/13/gstreamer-rtsp-server/
>
> Hope to get some advice ..
>
> --
> View this message in context:
> http://www.nabble.com/What-happens-when-set-always_copy%3DFALSE-in-v4l2src-tp23361935p23361935.html
> 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 <yangsb05 at gmail.com>
> Subject: [gst-devel] What happens when set always_copy=FALSE in
> v4l2src
> To: gstreamer-devel at lists.sourceforge.net
> Message-ID: <23361935.post at talk.nabble.com>
> 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
> http://blogs.gnome.org/uraeus/2008/10/13/gstreamer-rtsp-server/
>
> Hope to get some advice ..
>
> --
> View this message in context:
> http://www.nabble.com/What-happens-when-set-always_copy%3DFALSE-in-v4l2src-tp23361935p23361935.html
> 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 <virajk at gmail.com>
> Subject: [gst-devel] adder with nonblocking pads
> To: gstreamer-devel at lists.sourceforge.net
> Message-ID:
> <ecfff7140905032143g41c9a697t3a50b2effe1dc894 at mail.gmail.com>
> 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 <bilboed at gmail.com>
> Subject: Re: [gst-devel] adder with nonblocking pads
> To: Discussion of the development of GStreamer
> <gstreamer-devel at lists.sourceforge.net>
> Message-ID: <1241418356.722.2.camel at 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
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> > leaders in dedicated Performance & Operations tracks. Use code vel09scf
> > and Save an extra 15% before 5/3. http://p.sf.net/sfu/velocityconf
> > _______________________________________________ gstreamer-devel mailing
> list gstreamer-devel at lists.sourceforge.net
> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>
>
>
>
> ------------------------------
>
> Message: 5
> Date: Mon, 04 May 2009 10:04:06 +0200
> From: Sebastian Dr?ge <sebastian.droege at collabora.co.uk>
> Subject: Re: [gst-devel] Sync spectrum messages with audio being
> played?
> To: Discussion of the development of GStreamer
> <gstreamer-devel at lists.sourceforge.net>
> Message-ID: <1241424246.18835.59.camel at 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 --------------
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>
> ------------------------------
>
> Message: 6
> Date: Mon, 04 May 2009 12:00:25 +0200
> From: Nicolas Bertrand <nico at inattendu.org>
> Subject: Re: [gst-devel] How to save a stream buffer to a file ?
> To: Discussion of the development of GStreamer
> <gstreamer-devel at lists.sourceforge.net>
> Message-ID: <49FEBCB9.5070708 at inattendu.org>
> 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" <Vikas.Patel at imgtec.com>
> Subject: [gst-devel] GTK GUI+ gstreamer
> To: "Discussion of the development of GStreamer"
> <gstreamer-devel at lists.sourceforge.net>
> Message-ID:
> <C1A102308DC77D40BA4DFB4CF4C63BB2567726 at pumail1.pu.imgtec.org>
> 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
>
>
>
>
>
>
>
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> Register Now & Save for Velocity, the Web Performance & Operations
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> expert-led, hands-on workshops and two days of sessions from industry
> leaders in dedicated Performance & Operations tracks. Use code vel09scf
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>
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>
--
Sreerenj B
http://sreerenj.livejournal.com
bsreerenj at gmail.com
mob: +91 9995377714
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