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Hi,

> The only problem is the log is very big.

Log level can be set per category, like this:
--gst-debug playbin2:4

"gst-launch --gst-debug-help" lists up all the categories.
 see gst-launch-0.10 (1)

regards,
tskd
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On Wed, Feb 9, 2011 at 1:17 PM, Will Kelleher <wil...@nu...> wrote:
> Hi everyone,
>
> I've been trying to use my TI Davinci hardware as a USB webcam gadget.
>  I backported the webcam module to the 2.6.32 kernel and I can
> compile/load it successfully.  This module creates a v4l2 output
> device that can accept YUY2 and MJPEG formats.
>
> When I run
>
> gst-launch videotestsrc ! v4l2sink device=/dev/video1
>

so maybe I am missing something here, but if it is a camera/input
device, I think you probably want to use v4l2src..  since v4l2sink is
for output (tv tuners, or hw overlay on some platforms)

ie. something like: gst-launch v4l2src ! autovideosink

BR,
-R

> I get
>
> "ERROR: from element
> /GstPipeline:pipeline0/GstVideoTestSrc:videotestsrc0: Could not
> negotiate format"
>
> I've tried a variety of other pipelines but they all produce the same error.
>
> Does anyone have experience with sending output to this particular
> module (g_webcam)?  Am I doing something obviously wrong?
>
> Thanks,
>
> Will Kelleher
> Nuvixa, Inc.
>
> ------------------------------------------------------------------------------
> The ultimate all-in-one performance toolkit: Intel(R) Parallel Studio XE:
> Pinpoint memory and threading errors before they happen.
> Find and fix more than 250 security defects in the development cycle.
> Locate bottlenecks in serial and parallel code that limit performance.
> http://p.sf.net/sfu/intel-dev2devfeb
> _______________________________________________
> gstreamer-devel mailing list
> gst...@li...
> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>

Hi Tim & Topi,
Thanks, it worked.
The only problem is the log is very big.
How can I understand the flow of the data, events and things like that. what
I want is essentially an overview of the way the data flows, the no of
threads that get created, the allocation of buffers and buffer management
etc.
Regards
-- 
Vinod James

On Thu, Feb 10, 2011 at 6:29 PM, Tim-Philipp Müller <t....@ze...> wrote:

> On Wed, 2011-02-09 at 12:09 +0530, vinod james wrote:
>
>
> > I used gst-launch=0.10 and generated logs with GST_DEBUG set to 5.
> > I read that there is a playbin application which does auto type
> > finding and pipeline and graph building.
> > I want to generate a detailed log like gst-launch. I am not able to
> > generate it.
> > Please advise me how to generate the detailed log.
> > I am trying to understand the internals of gstreamer by looking at the
> > logs generated.
>
> playbin is not an application, it's an element/plugin. You can use it
> from gst-launch like this:
>
>  $ GST_DEBUG=*:5 gst-launch-0.10 playbin2 uri=file:///path/to/foo.avi
> 2>dbg.log
>
> Cheers
>  -Tim
>
>
>
>
>
> ------------------------------------------------------------------------------
> The ultimate all-in-one performance toolkit: Intel(R) Parallel Studio XE:
> Pinpoint memory and threading errors before they happen.
> Find and fix more than 250 security defects in the development cycle.
> Locate bottlenecks in serial and parallel code that limit performance.
> http://p.sf.net/sfu/intel-dev2devfeb
> _______________________________________________
> gstreamer-devel mailing list
> gst...@li...
> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>

I thought of set that configuration string but how could i set that, that is
huge data it started with AAAAX..... should i set with environment variable
and then set that or include whole thing in pipeline, i m confused please
help. Thank you.
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Hi,
On 02/10/2011 02:47 PM, vinod james wrote:
> can somebody please reply on this
>
> On Wed, Feb 9, 2011 at 12:09 PM, vinod james <vin...@gm...
> <mailto:vin...@gm...>> wrote:
>
>     Hi,
>     I am  a newbie to gtsreamer.
>     I went through the documentation on the website and cloned the
>     gstreamer and plugins and installed them.
>     I could test a couple of audios and videos successfully.
>
>     I used gst-launch=0.10 and generated logs with GST_DEBUG set to 5.
>
>     I read that there is a playbin application which does auto type
>     finding and pipeline and graph building.
>
>     I want to generate a detailed log like gst-launch. I am not able to
>     generate it.
>
>
>     Please advise me how to generate the detailed log.

Calling

gst-launch-0.10 --gst-debug=3 playbin2 uri=file://<path_to_your_file>

should give you a log that shows how playbin2 does things.

BR,
Topi

>
>     I am trying to understand the internals of gstreamer by looking at
>     the logs generated.
>
>
>     Thanks in Advance
>     --
>     Vinod James
>
>
>
>
> --
> Vinod James

On 02/10/2011 02:02 PM, Puneeth wrote:
>
> Hi All,
>
>           I m facing some problem when i  m using vorbis decoder with
> gstrtpbin with the below pipeline.
> Client
>         sudo gst-launch gstrtpbin name=rtpbin udpsrc
> caps="application/x-rtp,channels=1,media=(string)audio,clock-rate=(int)44100,encoding-name=(string)VORBIS,payload=(int)96,encoding-params=(string)1"
> port=5002 ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtpvorbisdepay ! vorbisdec !
> audioconvert ! volume volume=10 ! audioresample quality=10 ! alsasink udpsrc
> port=5003 ! rtpbin.recv_rtcp_sink_0 rtpbin.send_rtcp_src_0 ! udpsink
> port=5007 host=127.0.0.1 sync=false async=false
>

You are missing the configuration string in the RTP caps.

Wim

> WARNING: from element
> /GstPipeline:pipeline0/GstRtpVorbisDepay:rtpvorbisdepay0: Could not decode
> stream.
> Additional debug info:
> gstrtpvorbisdepay.c(584): gst_rtp_vorbis_depay_process ():
> /GstPipeline:pipeline0/GstRtpVorbisDepay:rtpvorbisdepay0:
> Could not switch codebooks
>
> Server
>
> sudo gst-launch gstrtpbin name=rtpbin alsasrc ! audioconvert ! audioresample
> ! vorbisenc ! rtpvorbispay ! rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 !
> udpsink port=5002 host=172.16.10.71 rtpbin.send_rtcp_src_0 ! udpsink
> port=5003 host=172.16.10.71 sync=false async=false udpsrc port=5007 !
> rtpbin.recv_rtcp_sink_0
>
> Please kindly anyone help me to resolve this issue. But i m able to listen
> with the below pipeline.
>
> gst-launch alsasrc ! audioconvert ! vorbisenc ! rtpvorbispay !
> rtpvorbisdepay ! vorbisdec ! audioconvert ! alsasink
>
> Thank u in advance.

Hi All,

         I m facing some problem when i  m using vorbis decoder with
gstrtpbin with the below pipeline.
Client 
       sudo gst-launch gstrtpbin name=rtpbin udpsrc
caps="application/x-rtp,channels=1,media=(string)audio,clock-rate=(int)44100,encoding-name=(string)VORBIS,payload=(int)96,encoding-params=(string)1"
port=5002 ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtpvorbisdepay ! vorbisdec !
audioconvert ! volume volume=10 ! audioresample quality=10 ! alsasink udpsrc
port=5003 ! rtpbin.recv_rtcp_sink_0 rtpbin.send_rtcp_src_0 ! udpsink
port=5007 host=127.0.0.1 sync=false async=false

WARNING: from element
/GstPipeline:pipeline0/GstRtpVorbisDepay:rtpvorbisdepay0: Could not decode
stream.
Additional debug info:
gstrtpvorbisdepay.c(584): gst_rtp_vorbis_depay_process ():
/GstPipeline:pipeline0/GstRtpVorbisDepay:rtpvorbisdepay0:
Could not switch codebooks

Server

sudo gst-launch gstrtpbin name=rtpbin alsasrc ! audioconvert ! audioresample
! vorbisenc ! rtpvorbispay ! rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 !
udpsink port=5002 host=172.16.10.71 rtpbin.send_rtcp_src_0 ! udpsink
port=5003 host=172.16.10.71 sync=false async=false udpsrc port=5007 !
rtpbin.recv_rtcp_sink_0

Please kindly anyone help me to resolve this issue. But i m able to listen
with the below pipeline.

gst-launch alsasrc ! audioconvert ! vorbisenc ! rtpvorbispay !
rtpvorbisdepay ! vorbisdec ! audioconvert ! alsasink

Thank u in advance.
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On Wed, 2011-02-09 at 12:09 +0530, vinod james wrote:

> I used gst-launch=0.10 and generated logs with GST_DEBUG set to 5.
> I read that there is a playbin application which does auto type
> finding and pipeline and graph building.
> I want to generate a detailed log like gst-launch. I am not able to
> generate it.
> Please advise me how to generate the detailed log.
> I am trying to understand the internals of gstreamer by looking at the
> logs generated.

playbin is not an application, it's an element/plugin. You can use it
from gst-launch like this:

 $ GST_DEBUG=*:5 gst-launch-0.10 playbin2 uri=file:///path/to/foo.avi
2>dbg.log

Cheers
 -Tim

can somebody please reply on this

On Wed, Feb 9, 2011 at 12:09 PM, vinod james <vin...@gm...> wrote:

> Hi,
> I am  a newbie to gtsreamer.
> I went through the documentation on the website and cloned the gstreamer
> and plugins and installed them.
> I could test a couple of audios and videos successfully.
>
> I used gst-launch=0.10 and generated logs with GST_DEBUG set to 5.
>
> I read that there is a playbin application which does auto type finding and
> pipeline and graph building.
>
> I want to generate a detailed log like gst-launch. I am not able to
> generate it.
>
>
> Please advise me how to generate the detailed log.
>
> I am trying to understand the internals of gstreamer by looking at the logs
> generated.
>
>
> Thanks in Advance
> --
> Vinod James
>

-- 
Vinod James

Sorry.
This issue was resolved.
I restored DEFAULT_RAW_CAPS definition in gst-plugins-base package.

Best Regards,
Hiroshi

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On Thu, Feb 10, 2011 at 09:15:49AM +0200, Marco Ballesio wrote:
> Hi,
> 
> On Wed, Feb 9, 2011 at 9:17 PM, Will Kelleher <wil...@nu...> wrote:
> > Hi everyone,
> >
> > I've been trying to use my TI Davinci hardware as a USB webcam gadget.
> >  I backported the webcam module to the 2.6.32 kernel and I can
> > compile/load it successfully.  This module creates a v4l2 output
> > device that can accept YUY2 and MJPEG formats.
> >
> > When I run
> >
> > gst-launch videotestsrc ! v4l2sink device=/dev/video1
> >
> > I get
> >
> > "ERROR: from element
> > /GstPipeline:pipeline0/GstVideoTestSrc:videotestsrc0: Could not
> > negotiate format"
> 
> probably you need to convert format/resolution/frame rate. If you want
> to know what went wrong with your caps negotiation, use -m as an
> option for gst-launch. If you want even more details, you can set
> GST_DEBUG=GST_CAPS:3 (or more).

You're much better off debugging caps negotiation problems using:

  GST_DEBUG=capsdebug:3 gst-launch videotestsrc ! capsdebug ! v4l2sink

David

Hi,

On Wed, Feb 9, 2011 at 9:17 PM, Will Kelleher <wil...@nu...> wrote:
> Hi everyone,
>
> I've been trying to use my TI Davinci hardware as a USB webcam gadget.
>  I backported the webcam module to the 2.6.32 kernel and I can
> compile/load it successfully.  This module creates a v4l2 output
> device that can accept YUY2 and MJPEG formats.
>
> When I run
>
> gst-launch videotestsrc ! v4l2sink device=/dev/video1
>
> I get
>
> "ERROR: from element
> /GstPipeline:pipeline0/GstVideoTestSrc:videotestsrc0: Could not
> negotiate format"

probably you need to convert format/resolution/frame rate. If you want
to know what went wrong with your caps negotiation, use -m as an
option for gst-launch. If you want even more details, you can set
GST_DEBUG=GST_CAPS:3 (or more).

A a last note, adding an ffmpegcolorspace ! videoscale ! videorate (or
a subet of the pipe) may help you.

Regard

>
> I've tried a variety of other pipelines but they all produce the same error.
>
> Does anyone have experience with sending output to this particular
> module (g_webcam)?  Am I doing something obviously wrong?
>
> Thanks,
>
> Will Kelleher
> Nuvixa, Inc.
>
> ------------------------------------------------------------------------------
> The ultimate all-in-one performance toolkit: Intel(R) Parallel Studio XE:
> Pinpoint memory and threading errors before they happen.
> Find and fix more than 250 security defects in the development cycle.
> Locate bottlenecks in serial and parallel code that limit performance.
> http://p.sf.net/sfu/intel-dev2devfeb
> _______________________________________________
> gstreamer-devel mailing list
> gst...@li...
> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>

I'm getting the following error, even after I added set_caps functions.
gst_video_format_get_component_width: assertion 'width > 0' failed.

My transform:

static GstFlowReturn
gst_xyz_transform_ip (GstBaseTransform * base, GstBuffer * outbuf) {
   Gstxyz *filter = GST_XYZ (base);

   if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (outbuf)))
     gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP 
(outbuf));

   if (filter->silent == FALSE)    g_print ("I'm in.\n");

   guint8 * data = GST_BUFFER_DATA (outbuf);

   GstFlowReturn ret = GST_FLOW_OK;
   VSImage dest = filter->dest;

// ERROR HERE
   gint lwidth  = gst_video_format_get_component_width (filter->format, 
0, filter->dest.width);

   gint lheight = 
240;//gst_video_format_get_component_height(filter->format, 0, 
filter->dest.height);
   gint offset = 1; //gst_video_format_get_component_offset 
(filter->format, 0, filter->dest.width, filter->dest.height);

   return ret;
}

Hi everyone,

I've been trying to use my TI Davinci hardware as a USB webcam gadget.
 I backported the webcam module to the 2.6.32 kernel and I can
compile/load it successfully.  This module creates a v4l2 output
device that can accept YUY2 and MJPEG formats.

When I run

gst-launch videotestsrc ! v4l2sink device=/dev/video1

I get

"ERROR: from element
/GstPipeline:pipeline0/GstVideoTestSrc:videotestsrc0: Could not
negotiate format"

I've tried a variety of other pipelines but they all produce the same error.

Does anyone have experience with sending output to this particular
module (g_webcam)?  Am I doing something obviously wrong?

Thanks,

Will Kelleher
Nuvixa, Inc.

I'm reading the plugin developer guide, chapter 4, and reading some 
sample plugins, in this case gstgamma.c.

In chapter 4, it says:
"In the element |_init ()| function, you create the pad from the pad 
template that has been registered with the element class in the 
|_base_init ()| function. After creating the pad, you have to set a 
|_setcaps ()| function pointer and optionally a |_getcaps ()| function 
pointer. Also, you have to set a |_chain ()| function pointer. 
Alternatively, pads can also operate in looping mode, which means that 
they can pull data themselves. More on this topic later. After that, you 
have to register the pad with the element. This happens like this:"

gstgamma sets a set_caps, transform_ip and before_transform functions, 
but does not set a chain function.
In my code that I started from boilerplate, set_caps is not set 
anywhere, but the plugin installs, and the trivial case displays video.  
As I progress, I'm finding some functionality that I assume doesn't work 
because I need to initialize the capability.  For example, calling 
gst_video_format_get_component_width() complains because it says the 
format doesn't match.

I suppose I'm asking if these functions get added by default, and 
second, should I take the writers guide as truth, or more of a suggestion?

Hi,
I'm new of Gstreamer. I've to play a file audio. I've three outputs and I
have to choose on which output I have to play the audio.
How can I do?
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Hello,
I have a question about gstreamer-vaapi plug-in.

I added "video/x-vaapi-surface" to DEFAULT_RAW_CAPS in gst-plugins-base
package.
And I did the following command and it played.
> gst-launch -v filesrc location=/home/me/test.mpeg ! qtdemux ! decodebin2 !
> vaapisink

Next, I tried to do the following command.
> gst-launch playbin2 uri=file:/home/me/test.mpeg

But the movie does not play. (The movie format is mpeg-4.)
By the log of Gstreamer, I confirmed that vaapidecode element was loaded.
But vaapisink element was not loaded. I want to know why vaapisink was not
loaded.
Please teach me.

I installed the following packages,
gstreamer:0.10.29
gstreamer plugins base: 0.10.28
gstreamer plugins good: 0.10.16
gstreamer vaapi: 0.2.5

Best Regards,
Wada
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Hi,
I have avertv hybrid volar hx capture card. The video format is composite
NTSC. Im able to play video with v4l2src ! xvimagesink pipeline. 
I want to convert the composite format to digital video format(Ex 264 or
ogg).
How do i do it using gstreamer. 

Thanx
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On Wed, Feb 09, 2011 at 01:43:16AM -0800, marcel.tella wrote:
> 
> Hi!
> 
> I need to stream in a webpage. In order to do this, I need to stream not an
> UDP stream, I think I need an HTTP stream, because browser's doesn't support
> other protocols... and firewalls don't like rtp usually.
> 
> Is it right? So, Is there some other possibility to stream to the web, to a
> HTML 5 page?
> 
> I've thought also in the apple's http live streaming, but I really don't
> know if I can use something like this in gstreamer.
> 
> I'm a bit lost, all the information you can give me will be welcome, so,
> thank you very much.
> 
> 

You can try using icecast2. GStreamer can be used as an icecast2 source
using the shout2send element. One thing to consider is the supported
formats and codecs. We currently use Ogg+Theora+Vorbis. The tool we use
to capture, encode and send the video is Landell
(http://landell.holoscopio.com/).

Regards,
Cascardo.

Hi Marcel,

Did you have a view on Flumotion? It's a HTTP streaming server based on
Gstreamer! Checkout http://www.flumotion.net/ please.

Frans van Berckel

> Hi!
>
> I need to stream in a webpage. In order to do this, I need to stream not
> an UDP stream, I think I need an HTTP stream, because browser's doesn't
> support other protocols... and firewalls don't like rtp usually.

> Is it right? So, Is there some other possibility to stream to the web, to
> a
> HTML 5 page?

> I've thought also in the apple's http live streaming, but I really don't
> know if I can use something like this in gstreamer.

> I'm a bit lost, all the information you can give me will be welcome, so,
> thank you very much.

Hi all

I have add to the pipeline the sync_bus_handler function, to change
the priority of the task. As I see from top there are a lot of thread created
by gstreamer but the CREATE status is called just for two threads, do you
know the reason?

Michael Trimarchi

Hi!

I need to stream in a webpage. In order to do this, I need to stream not an
UDP stream, I think I need an HTTP stream, because browser's doesn't support
other protocols... and firewalls don't like rtp usually.

Is it right? So, Is there some other possibility to stream to the web, to a
HTML 5 page?

I've thought also in the apple's http live streaming, but I really don't
know if I can use something like this in gstreamer.

I'm a bit lost, all the information you can give me will be welcome, so,
thank you very much.

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Hi,

On Wed, Feb 9, 2011 at 1:41 AM, Thierry Panthier
<thi...@gm...> wrote:
> Hi Marco,
>
> Sorry I didn't make myself clear. You were right I'm actually looking
> for a way to handle the RTCP timeout for a specific SSRC.
>
>> as you're likely using UDP as transport layer, RTP packets cannot be
>> timed out (yes, it's an unreliable protocol). RTCP can help you here,
>> as you'd just need to enable it and listen for (missing) Sender
>> Reports, translated in "on-ssrc-active" signals from the session
>> element in the GstRtspSrc (which is usually a GstRtpBin).
>
> The problem is that the rtspsrc bin does not have this signal. And if
> I try to connect to it I get:
>
>       TypeError: <__main__.GstRTSPSrc object (rtspsrc0) at
> 0x919beb4>: unknown signal name: on-ssrc-active
>
> The rtspsrc documentation says it is built on top of gstrtpbin but how
> do I get access to it?

it's not possible to get such a signal from the rtspsrc, which is just
a specialised bin (see here to believe:
http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-rtspsrc.html).

As you already suspect, it's instead possible to get such a signal
from the GstRtpBin used as a manager (I'm sorry I mistakenly wrote
"session" earlier, but it's actually possible to do the same with the
GstRtpSession element) inside the RtspSrc. You can get it through the
GstBin api:

http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBin.html

Basically, iterate through the children and get the one instantiating
a GstRtpBin. A more quick-and-dirty way would be to use
gst_bin_get_by_name and something like "rtpbin0" for the name.

Regards

>
> If could simply have access to the signals "on-ssrc-active" and
> "on-timeout" that would be great.
>
> However if the implementation does not allow me to do that then I
> would like to know if it is possible to add an element to my bin to
> detect a "frozen stream". I've gone through all the options given by
> gst-inspect but couldn't find anything useful.
>
>> RTCP packets are (usually) sent with intervals of 5s. If you want
>> something faster, you can install a data probe somewhere in the pipe
>> resetting a timeout each time a buffer transits through the pad. When
>> the timer triggers, then a timeout occurred and you can unilaterally
>> terminate the communication.
>
> 5s for me it's good enough for my application.
>
>
> Thanks in advance,
>
> Thierry
>
> ------------------------------------------------------------------------------
> The ultimate all-in-one performance toolkit: Intel(R) Parallel Studio XE:
> Pinpoint memory and threading errors before they happen.
> Find and fix more than 250 security defects in the development cycle.
> Locate bottlenecks in serial and parallel code that limit performance.
> http://p.sf.net/sfu/intel-dev2devfeb
> _______________________________________________
> gstreamer-devel mailing list
> gst...@li...
> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>

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2001	Jan (137)	Feb (126)	Mar (247)	Apr (140)	May (232)	Jun (136)	Jul (61)	Aug (135)	Sep (135)	Oct (113)	Nov (46)	Dec (157)
2002	Jan (139)	Feb (127)	Mar (153)	Apr (174)	May (95)	Jun (115)	Jul (202)	Aug (123)	Sep (208)	Oct (114)	Nov (123)	Dec (91)
2003	Jan (296)	Feb (119)	Mar (112)	Apr (235)	May (205)	Jun (271)	Jul (219)	Aug (104)	Sep (149)	Oct (200)	Nov (242)	Dec (466)
2004	Jan (447)	Feb (300)	Mar (485)	Apr (267)	May (205)	Jun (183)	Jul (344)	Aug (176)	Sep (119)	Oct (140)	Nov (154)	Dec (152)
2005	Jan (209)	Feb (178)	Mar (128)	Apr (166)	May (163)	Jun (150)	Jul (191)	Aug (166)	Sep (212)	Oct (212)	Nov (240)	Dec (236)
2006	Jan (178)	Feb (184)	Mar (188)	Apr (189)	May (267)	Jun (198)	Jul (151)	Aug (212)	Sep (190)	Oct (180)	Nov (354)	Dec (199)
2007	Jan (211)	Feb (173)	Mar (182)	Apr (151)	May (233)	Jun (288)	Jul (213)	Aug (221)	Sep (320)	Oct (301)	Nov (193)	Dec (214)
2008	Jan (235)	Feb (254)	Mar (237)	Apr (232)	May (187)	Jun (239)	Jul (353)	Aug (362)	Sep (431)	Oct (423)	Nov (358)	Dec (351)
2009	Jan (408)	Feb (377)	Mar (547)	Apr (437)	May (483)	Jun (449)	Jul (309)	Aug (297)	Sep (279)	Oct (329)	Nov (336)	Dec (290)
2010	Jan (237)	Feb (296)	Mar (523)	Apr (515)	May (340)	Jun (474)	Jul (372)	Aug (427)	Sep (343)	Oct (396)	Nov (407)	Dec (512)
2011	Jan (515)	Feb (146)	Mar	Apr	May	Jun	Jul	Aug	Sep	Oct	Nov	Dec