gmane.comp.video.gstreamer.devel

Re: How to save incoming h264 stream without re-encoding?

W.A. Garrett Weaver — 2012-05-25T03:54:09

I found the solution to the intermittent delay problem. What you have to do
is reduce the latency on gstrtpjitterbuffer to 10ms instead of being 500ms.

gst-launch-0.10 udpsrc multicast-group=224.1.1.1 auto-multicast=true
port=5010 caps='application/x-rtp, media=(string)video,
clock-rate=(int)90000, encoding-name=(string)H264,
sprop-parameter-sets=(string)\"Z0KAHukBQHpCAAAH0AAB1MAIAA\\=\\=\\,aM48gAA\\=\",
payload=(int)96, ssrc=(uint)3315029550, clock-base=(uint)3926529534,
seqnum-base=(uint)45576' ! gstrtpjitterbuffer drop-on-latency=true
latency=10 ! rtph264depay ! ffdec_h264 ! x264enc ! matroskamux ! filesink
location=movie.mkv

The result is this:
http://www.youtube.com/watch?v=br--9h3-g4U

This works good if the computer on the receiving end is sufficiently fast
enough to compress video, since mine is, this is an adequate solution.
Still, it would be nice to record the h264 stream directly to a file
instead of having to re compress.

On Wed, May 16, 2012 at 2:01 PM, W.A. Garrett Weaver <
weaverg< at >email.arizona.edu> wrote:

can't control valve drop property

jomifo — 2012-05-24T19:41:16

I am mixing a video stream with a static image.  That's working fine.  I
would like to dynamically turn the image off/on at certain times.  My idea
was to put a valve after the image stream and use a controller to toggle the
drop property.  But alas, I get an error when trying to create that
controller,

In [45]: gst.Controller( gst.element_factory_make( 'valve' ), 'drop' )

** (ipython:2530): CRITICAL **: gst_controlled_property_new: assertion
`(pspec->flags & (G_PARAM_WRITABLE | GST_PARAM_CONTROLLABLE |
G_PARAM_CONSTRUCT_ONLY)) == (G_PARAM_WRITABLE | GST_PARAM_CONTROLLABLE)'
failed
---------------------------------------------------------------------------
RuntimeError                              Traceback (most recent call last)

/mnt/transfer/<ipython console> in <module>()

RuntimeError: could not create GstController object

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Re: How to run a 2-to-1 element

iron_guitarist1987 — 2012-05-24T15:22:31

Is the code even correct? It compiles and installs, but I haven't been able
to test it (don't know how).

Anyone? 

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Re: Streaming 1080p24 H264/AVC and MPEG4/AAC, lip synchronizationusing gstrtpbin, unable to link in video_00? (bug?)

Martin Lund — 2012-05-24T14:51:45

Hi Wim,

Thank you for the speedy response. Latest git does indeed seem to fix
this particular issue.

Now the server pipelines seems to play correctly. Next, I need to
configure and verify the clients.

/Martin

On Thu, May 24, 2012 at 3:42 PM, Martin Lund <martin.lund< at >ixonos.com> wrote:

Re: Streaming 1080p24 H264/AVC and MPEG4/AAC, lip synchronizationusing gstrtpbin, unable to link in video_00? (bug?)

Martin Lund — 2012-05-24T13:42:11

Hi,

I'll have to take your word for it since I'm not familiar with the
gstreamer internals.

I'll try to build and test latest from git to see how that works out.

For now, is there possibly a way to bypass this bug in my gst-launch command?

The bug thread you are referring to talks about bypass by forcing
capsfilters but I assume this is meant for the client side so I'm not
sure it applies to my server side of things?

/Martin

On Thu, May 24, 2012 at 3:14 PM, Wim Taymans <wim.taymans< at >gmail.com> wrote:

Re: Streaming 1080p24 H264/AVC and MPEG4/AAC, lip synchronizationusing gstrtpbin, unable to link in video_00? (bug?)

Martin Lund — 2012-05-24T13:16:14

I'm sorry about the formatting of my previous mail. Here is my
gst-launch command in hopefully more readable layout:

gst-launch --gst-debug=4 --verbose --gst-debug-no-color \
    gstrtpbin name=rtpbin \
        filesrc location=battleship.mp4 ! qtdemux name=demux \
            demux.video_00 ! queue ! rtph264pay ! rtpbin.send_rtp_sink_0 \
                  rtpbin.send_rtp_src_0 ! udpsink port=5000 \
                  rtpbin.send_rtcp_src_0 ! udpsink port=5001
sync=false async=false \
                  udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \
            demux.audio_00 ! queue ! rtpmp4apay ! rtpbin.send_rtp_sink_1 \
                  rtpbin.send_rtp_src_1 ! udpsink port=5002 \
                  rtpbin.send_rtcp_src_1 ! udpsink port=5003
sync=false async=false \
                  udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1

On Thu, May 24, 2012 at 3:02 PM, Martin Lund <martin.lund< at >ixonos.com> wrote:

Re: Streaming 1080p24 H264/AVC and MPEG4/AAC, lip synchronizationusing gstrtpbin, unable to link in video_00? (bug?)

Wim Taymans — 2012-05-24T13:14:17

It sounds like https://bugzilla.gnome.org/show_bug.cgi?id=672019 that 
was just fixed.

Wim

Streaming 1080p24 H264/AVC and MPEG4/AAC, lip synchronization usinggstrtpbin, unable to link in video_00? (bug?)

Martin Lund — 2012-05-24T13:02:41

Hi,

I'm on a mission to have a server stream a file containing H264/AVC
and MPEG4/AAC to two other hosts. One of these hosts will play the
video stream and the other will play the audio stream. I need A/V lip
syncrhonization between these two hosts so as far as I understand I
need to use the gstrtpbin plugin in order to utilize the RTP/RTCP
protocol to obtain this kind of synchronization.

I'm playing around with gst-launch to test the RTP/RTCP stuff. My
server pipeline is based on the RTP reference example found here:
http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh

This is the basic server pipeline I'm trying to configure (assumes it
all running on the same server, ie. no "host=" stuff):

gst-launch
                \
    gstrtpbin name=rtpbin
                \
        filesrc location=battleship.mp4 ! qtdemux name=demux
                \
            demux.video_00 ! queue ! rtph264pay !
rtpbin.send_rtp_sink_0               \
                  rtpbin.send_rtp_src_0 ! udpsink port=5000
                \
                  rtpbin.send_rtcp_src_0 ! udpsink port=5001
sync=false async=false    \
                  udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0
                \
            demux.audio_00 ! queue ! rtpmp4apay !
rtpbin.send_rtp_sink_1               \
                  rtpbin.send_rtp_src_1 ! udpsink port=5002
                \
                  rtpbin.send_rtcp_src_1 ! udpsink port=5003
sync=false async=false    \
                  udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1

This fails to link in the video stream?

With debug enabled it spews out various errors regarding
demux:video_00 and queue0 unable to connect:

GST_ELEMENT_PADS gstutils.c:1101:gst_element_get_compatible_pad:
finding pad in queue0 compatible with demux:video_00
GST_PADS gstutils.c:1032:gst_pad_check_link: trying to link
demux:video_00 and queue0:sink
GST_CAPS gstpad.c:2336:gst_pad_get_caps_reffed:<demux:video_00> get pad caps
GST_CAPS gstpad.c:2246:gst_pad_get_caps_unlocked:<demux:video_00> get pad caps
GST_CAPS gstpad.c:2250:gst_pad_get_caps_unlocked:<demux:video_00>
dispatching to pad getcaps function
GST_CAPS gstpad.c:2263:gst_pad_get_caps_unlocked:<demux:video_00> pad
getcaps returned video/x-h264, stream-format=(string)avc,
alignment=(string)au, level=(string)4, profile=(string)high,
codec_data=(buffer)01640028ffe1001c67640028acd94078067b016a020202800000030080015f90078c18cb01000668ebe08cb22c,
width=(int)1920, height=(int)816, framerate=(fraction)45000/1877,
pixel-aspect-ratio=(fraction)1/1
GST_ELEMENT_PADS
gstutils.c:1195:gst_element_get_compatible_pad:<queue0> Could not find
a compatible unlinked always pad to link to demux:video_00, now
checking request pads
...
GST_ELEMENT_PADS gstutils.c:1829:gst_element_link_pads_full: no link
possible from demux:video_00 to queue0
default gstutils.c:2037:gst_element_link_pads_filtered: Could not link
pads: demux:video_00 - queue0:(null)


Yet, the same troublesome pipeline elements is tested to work
perfectly for direct streaming when not in combination with gstrtpbin:

gst-launch filesrc location=battleship.mp4 ! qtdemux name=demux \
demux.video_00 ! queue ! rtph264pay ! udpsink port=3000 \
demux.audio_00 ! queue ! rtpmp4apay ! udpsink port=5000

So obviously demux:video_00 and queue0:sink can actually connect.

What I don't understand why my server pipeline does not work in
combination with gstrtpbin... Might this be a bug or am I missing
something here?

I'm testing on Ubuntu 12.04, gstreamer: base 0.10.36, good 0.10.31,
bad 0.10.22.3.

Any help/input is appreciated.

Br, Martin

P.s. The clip I'm using is found here
http://download4.dvdloc8.com/trailers/divxdigest/battleship-trailer2.zip

Re: send buffer through udpsink

enricom — 2012-05-24T09:56:10

Hello, to do so first of all you need to create the buffer, for instance
with: buffer  =  gst_buffer_new_and_alloc (sizeof(your_data)): this way you
create a buffer with required size. Then you have to put your data into it
by using memcpy(GST_BUFFER_DATA(buffer) , your_data, sizeof(your_data)). 
After this step you need to inject it into your pipeline by using an appsrc
element, with gst_app_src_push_buffer(GST_APP_SRC(appsrc), buffer)). Of
course appsrc needs to be connected to the udpsink element. On the other
side you can use udpsrc, then an identity element. This one, by means of the
handoff signal, tells you when a buffer has been received and so you can
extract data from it.





dennis wrote


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ffenc_msmpeg4v2: failed to encode buffer

padam — 2012-05-24T06:49:27

Hi, reffering to 
http://gstreamer-devel.966125.n4.nabble.com/Gnonlin-question-extract-audio-video-clip-td2305307.html#a3063249
THIS  topic i created this pipe:

gst-launch gnlfilesource name=video
caps="video/x-raw-yuv,width=320,height=240" location=file:///D:\\source.avi
start=0 duration=23000000000 media-start=1188000000
media-duration=23000000000 ! identity single-segment=true ! progressreport
update-freq=1 ! ffmpegcolorspace ! ffenc_msmpeg4v2 bitrate=350000 ! avimux
name=mux ! filesink location=D:\\destination.avi gnlfilesource name=audio
caps="audio/x-raw-int,rate=16000,channels=1" location=file:///D:\\source.avi
start=0 duration=23000000000 media-start=1188000000
media-duration=23000000000 ! identity single-segment=true ! audioconvert !
lamemp3enc bitrate=32 ! mux.

When i launch this sometimes, on the same file, i get that errors:

0:00:11.781250000  2512   00B5F320 ERROR                 ffmpeg .:0:: Error,
Inv
alid timestamp=0, last=0
0:00:11.781250000  2512   00B5F320 ERROR                 ffmpeg
gstffmpegenc.c:7
23:gst_ffmpegenc_chain_video:<ffenc_msmpeg4v20> ffenc_msmpeg4v2: failed to
encode buffer

I have read that i should add to my pipe single-segment=true, but i already
have it in my pipe, so where is the problem?

And another problem is that i would like to set new properties for
destination.avi, but gstreamer seems to ignoring all my properties... Why is
that?

thanks for help

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Re: How to list elements used by playbin2?

Kyrylo V Polezhaiev — 2012-05-23T20:59:15

Thank you very much! :)

Re: How to list elements used by playbin2?

Mart Raudsepp — 2012-05-23T20:14:27

You could look at a debug log with GST_DEBUG=GST_ELEMENT_FACTORY:3 in
your enviroment to see which elements are created.
However, you need to keep in mind that playbin2 does auto-plugging of
elements based on the media container, used video/audio codec and so on,
so by including elements purely based on such metrics, only the same
kind of media files will be possible to be played then. This may be fine
if your application only uses GStreamer to play also included known
media, not user chosen. Otherwise you should probably take a different
approach in choosing which element plugins to include.

Re: no such pad 'video_%04x' in element "tsdemux"

Maxime Louvel — 2012-05-23T20:00:58

thanks Emile,
it works

On Wed, May 23, 2012 at 12:39 PM, Emile Semmes <emile.semmes< at >e6group.com>wrote:

How to list elements used by playbin2?

Kyrylo V Polezhaiev — 2012-05-23T19:16:12

Hello, I want to insert several dlls in my software installer.
So, I need to know, what plugins to include. My program uses playbin2 and two other gstreamer elements.
How can I ask playbin2 to list the elements he used?

Re: no such pad 'video_%04x' in element "tsdemux"

Emile Semmes — 2012-05-23T17:39:28

Hi Maxime,

For mpegtsdemux, you can't link the src pads until they are added. This 
happens during runtime so you'll need to add a signal handler on the 
pad-added signal from the mpegtsdemux element. Take a look at section 
8.1 of the documentation 
(http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/chapter-pads.html) 
for an example on how this is done. Section 8.1.1 uses oggdemux but the 
method is the same. In your callback, you can link your pad there. Pay 
attention to the caps since you don't until later if it's an audio or 
video track.

HTH,

Emile

--
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Software Consultant
e6 Group, LLC
Office: (630) 376-0626
www.e6group.com



On 5/22/2012 8:56 AM, Maxime Louvel wrote:

Re: Building a stand-alone, static, debug'-g' version of GStreamer

FritzKatz — 2012-05-23T18:24:26

Thank you Luis!  Works great!

For future reference, anyone google-searching into this thread that are
behind a firewall like me will need to modify the gst-uninstalled scripts to
do "http-dumb-git". For example:
   
    $ git clone http://anongit.freedesktop.org/git/gstreamer/gstreamer.git 

and
    gstreamer.doap:     
        <location
rdf:resource="http://anongit.freedesktop.org/gstreamer/gstreamer.git"/>

instead of:
    gstreamer.doap_ORIG:     
        <location
rdf:resource="git://anongit.freedesktop.org/gstreamer/gstreamer"/>





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Re: rtpjitterbuffer and percent property

Daniel Mellado — 2012-05-23T16:39:44

I meant in the pipeline:

i.e rtspsrc buffer-mode=0 latency=0 ! gstrtpjitterbuffer mode=2 latency=4000
! decodebin2 ! autovidesink

I get messages by parsing buffer level until 100% then video plays, but if
the network it's down, and the buffer is used, I don't get any info from
percentage goiing fro 100 to 16 percent, it just suddenly drops.

ie -> percent 100%
next message
-> percent 16%

Is there any way I can get that info? (values from percent property when
playing from buffer)
ie.

Buffer 0%---messages 0.1...100% (Up to here it's ok)-->100%..messages from
100 to 16..16%.

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Re: rtpjitterbuffer and percent property

Daniel Mellado — 2012-05-23T16:29:43

Tried using a jitterbuffer not inside rtsprc but in the queue, and the
behaviour it's the same. Even accessing the percent property or messages,
They go from 0 to 100%, then suddenly drops to 16% (low threshold in
gstrtpjitterbuffer).

¿How can I access the intermediate values?



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Receive RTSP stream without decoding - flumotion

Hossam Khankan — 2012-05-23T15:24:24

rtsp-producer is a component in flumotion Streaming Server (built on
gstreamer) which recives RTSP stream, decode it and serves decoded audio
and video to the next component in the flow.
I want to modify this component to skip decoding; I just want to recive
RTSP stream and expose the encoded A/V (audio:aac/video:h264) to the next
component (which will mux them useing mp4mux before serving them through
http-streamer to HTML5 webpage)
This is the RTSP component code file (rtsp.py) which generates the pipeline:

class Rtsp(feedcomponent.ParseLaunchComponent):

    def get_pipeline_string(self, properties):
        width = properties.get('width', 0)
        height = properties.get('height', 0)
        location = properties.get('location')
        framerate = properties.get('framerate', (25, 2))
        has_audio = properties.get('has-audio', True)
        if width > 0 and height > 0:
            scaling_template = (" videoscale method=1 ! "
                "video/x-raw-yuv,width=%d,height=%d !" % (width, height))
        else:
            scaling_template = ""
        if has_audio:
            audio_template = "d. ! queue ! audioconvert ! audio/x-raw-int"
        else:
            audio_template = "fakesrc silent=true ! audio/x-raw-int"
        return ("rtspsrc name=src location=%s ! decodebin name=d ! queue "
                " ! %s ffmpegcolorspace ! video/x-raw-yuv "
                " ! videorate ! video/x-raw-yuv,framerate=%d/%d ! "
                " < at >feeder:video< at > %s ! < at >feeder:audio< at >"
                % (location, scaling_template, framerate[0],
                   framerate[1], audio_template))

How do I modify this to achieve my goal ?
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Re: How to play IPTV streamer by udpsrc？

tanzhichaoanuo — 2012-05-22T13:52:25

Hello all:
 I have resolved the above problem.
 The IPTV multicast package can't be received and played by gstreamer,this
is because udpsrc plugin can't get the net stream,but  when I debug in the
kernel,I found the multicast package was dropped in the IP layer because of
IP routing.Why will result this？  This is because the source address of the
IPTV stream.
for example:
I use tcpdump -i eth0 ,and get the follow infor: 
09:11:55.138926 IP 61.181.150.102.12502 > 225.1.2.2.8034: UDP, length 1348 
09:11:55.142901 IP 61.181.150.102.12502 > 225.1.2.2.8034: UDP, length 1348 
09:11:55.146890 IP 61.181.150.102.12502 > 225.1.2.2.8034: UDP, length 1348 
09:11:55.150871 IP 61.181.150.102.12502 > 225.1.2.2.8034: UDP, length 1348 
09:11:55.154911 IP 61.181.150.102.12502 > 225.1.2.2.8034: UDP, length 1348 
09:11:55.160492 IP 61.181.150.102.12502 > 225.1.2.2.8034: UDP, length 1348 
09:11:55.164337 IP 61.181.150.102.12502 > 225.1.2.2.8034: UDP, length 1348 

the source IP addresss was 61.181.150.102,and my local IP address was
192.168.0.100.and the local route table don't include 61.XX.XX.XX
After reconfig the kernel,and select the net options ,reconfig the IP route
infor will resolve this.
 Thanks all the same. 
                             zhichao. 


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How to run a 2-to-1 element

iron_guitarist1987 — 2012-05-23T14:32:52

I have [finally] created a 2-to-1 simple element. My question now is how to
run it (an example gst-launch using 2 videotestsrc and 1 autosink). Here is
the code:



// gst_ntoone.c - Notes on creating a ntoone

/**
 * SECTION:element-plugin
 *
 * This plugin serves as a base template and a learning tool
 * for creating chain elements. All it does is to 
 *
 * <refsect2>
 * <title>Example launch line</title>
 * |[
 * gst-launch -v -m videontoonesrc pattern=snow ! ntoone ! autovideosink
 * ]|
 * </refsect2>
 */

#ifdef HAVE_CONFIG_H
#  include <config.h>
#endif

#include <gst/gst.h>
#include "gstntoone.h"
#include <opencv/cv.h>
#include <opencv/highgui.h>

GST_DEBUG_CATEGORY_STATIC (gst_ntoone_debug);
#define GST_CAT_DEFAULT gst_ntoone_debug

enum
{
  PROP_0,
  PROP_SILENT,
  LINE_COLOR
};

/* the capabilities of the inputs and outputs.
 *
 * describe the real formats here.
 */

//Creates a template for the pads. In the _init() function, you can create
//as many pads you want from these templates.
static GstStaticPadTemplate video_sink_factory = GST_STATIC_PAD_TEMPLATE
("video_sink",
    GST_PAD_SINK,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("ANY")
    );

static GstStaticPadTemplate klv_sink_factory = GST_STATIC_PAD_TEMPLATE
("klv_sink",
    GST_PAD_SINK,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("ANY")
    );

static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
    GST_PAD_SRC,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("ANY")
    );

GST_BOILERPLATE (GstNtoone, gst_ntoone, GstElement,
    GST_TYPE_ELEMENT);

//function prototypes
static void gst_ntoone_set_property (GObject * object, guint prop_id,
    const GValue * value, GParamSpec * pspec);
static void gst_ntoone_get_property (GObject * object, guint prop_id,
    GValue * value, GParamSpec * pspec);
static gboolean gst_ntoone_set_caps (GstPad * pad, GstCaps * caps);
static GstFlowReturn gst_ntoone_collected (GstCollectPads * pads, GstNtoone
* filter);

/* GObject vmethod implementations */
static void
gst_ntoone_base_init (gpointer gclass)
{
  GstElementClass *element_class = GST_ELEMENT_CLASS (gclass);

  //Describe the element's details
  //    Plugin name
  //    Plugin type
  //    A brief description
  //    Author and email (email is optional)
  gst_element_class_set_details_simple(element_class,
    "Plugin Template",
    "Ntoone",
    "Generic Chain Element",
    "Jason Trinidad jtrinidad< at >eoir.com");

  
  //Register the tamplates. They can be used
  //in the init() function to create pads
  gst_element_class_add_pad_template (element_class,
      gst_static_pad_template_get (&src_factory));
  gst_element_class_add_pad_template (element_class,
      gst_static_pad_template_get (&video_sink_factory));
  gst_element_class_add_pad_template (element_class,
      gst_static_pad_template_get (&klv_sink_factory));
}

/* initialize the plugin's class */
static void
gst_ntoone_class_init (GstNtooneClass * klass)
{
  GObjectClass *gobject_class;
  GstElementClass *gstelement_class;

  gobject_class = (GObjectClass *) klass;
  gstelement_class = (GstElementClass *) klass;

  gobject_class->set_property = gst_ntoone_set_property;
  gobject_class->get_property = gst_ntoone_get_property;

  g_object_class_install_property (gobject_class, PROP_SILENT, 
  g_param_spec_boolean ("silent", "Silent", "Produce verbose output
?",FALSE, G_PARAM_READWRITE));
  

  g_object_class_install_property (gobject_class, LINE_COLOR, 
  g_param_spec_string ("line_color", "Line_color", "Chenge the color of the
line", "red", G_PARAM_READWRITE));

}

/* initialize the new element
 * instantiate pads and add them to element
 */
static void
gst_ntoone_init (GstNtoone * filter,
    GstNtooneClass * gclass)
{
  filter->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
  gst_pad_set_getcaps_function (filter->srcpad,
                                GST_DEBUG_FUNCPTR(gst_pad_proxy_getcaps));
  
  filter->sinkpad1 = gst_pad_new_from_static_template (&video_sink_factory,
"video_sink");
  gst_pad_set_setcaps_function (filter->sinkpad1,
                                GST_DEBUG_FUNCPTR(gst_ntoone_set_caps));
  gst_pad_set_getcaps_function (filter->sinkpad1,
                                GST_DEBUG_FUNCPTR(gst_pad_proxy_getcaps));
  
  filter->sinkpad2 = gst_pad_new_from_static_template (&klv_sink_factory,
"klv_sink");
  gst_pad_set_setcaps_function (filter->sinkpad2,
                                GST_DEBUG_FUNCPTR(gst_ntoone_set_caps));
  gst_pad_set_getcaps_function (filter->sinkpad2,
                                GST_DEBUG_FUNCPTR(gst_pad_proxy_getcaps));
  

  filter->collect = gst_collect_pads_new ();
  gst_collect_pads_set_function (filter->collect,
      (GstCollectPadsFunction) gst_ntoone_collected, filter);
  
  gst_collect_pads_add_pad (filter->collect, filter->sinkpad1, sizeof
(GstCollectData));
  gst_collect_pads_add_pad (filter->collect, filter->sinkpad2, sizeof
(GstCollectData));
  
  gst_element_add_pad (GST_ELEMENT (filter), filter->sinkpad1);
  gst_element_add_pad (GST_ELEMENT (filter), filter->sinkpad2);
  gst_element_add_pad (GST_ELEMENT (filter), filter->srcpad);
  filter->silent = FALSE;
  
}

/*-------------------------------------------------------------------
 * _set_property() is used to set arguments in the element.
 * they can be used when running a pipelin by just typing the
 * property name and the value right next to the plugin
 * e.g. gst-launch -v -m videontoonesrc pattern=snow ! ntoone
line_color=green ! autovideosink
 * where pattern and line_color are properties
 -------------------------------------------------------------------*/
static void
gst_ntoone_set_property (GObject * object, guint prop_id,
    const GValue * value, GParamSpec * pspec)
{
  GstNtoone *filter = GST_NTOONE (object);
   
  switch (prop_id) {
    case PROP_SILENT:
      filter->silent = g_value_get_boolean (value);
      break;
    case LINE_COLOR:
      g_free (filter->line_color); 
      filter->line_color = g_value_dup_string (value); 
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static void
gst_ntoone_get_property (GObject * object, guint prop_id,
    GValue * value, GParamSpec * pspec)
{
  GstNtoone *filter = GST_NTOONE (object);

 
  switch (prop_id) {
    case PROP_SILENT:
      g_value_set_boolean (value, filter->silent);
      break;
    case LINE_COLOR:
      g_value_set_string (value, filter->line_color);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
  
}

/* GstElement vmethod implementations */

/* this function handles the link with other elements */
static gboolean
gst_ntoone_set_caps (GstPad * pad, GstCaps * caps)
{
  GstStructure *structure = gst_caps_get_structure(caps,0);
  GstNtoone *filter;
  GstPad *otherpad;

  filter = GST_NTOONE (gst_pad_get_parent (pad));
  
  gst_structure_get_int (structure, "rate", &filter->srcpad);
  otherpad = (pad == filter->srcpad) ? filter->sinkpad1 : filter->srcpad;
  gst_object_unref (filter);

  
  return gst_pad_set_caps (otherpad, caps);
}


static GstFlowReturn
gst_ntoone_collected (GstCollectPads * pads, GstNtoone * filter)
{
  guint size;
  GstCollectData *cdata;
  GstBuffer *outbuf, *sink1buf, *sink2buf;
  GstFlowReturn ret = GST_FLOW_OK;
  GSList *collected;
  guint nsamples;
  guint ncollected = 0;
  gboolean empty = TRUE;
  
  size = gst_collect_pads_available (pads); //Query how much bytes can be
read from each queued buffer. 
     //This means that the result of this call is the maximum 
     //number of bytes that can be read from each of the pads.
  
  GST_DEBUG_OBJECT (filter, "Starting to collect %u bytes", size);
  
  collected = pads->data;
  cdata = (GstCollectData *) collected->data;
  sink1buf = gst_collect_pads_take_buffer (pads, cdata, size);
  
  collected = collected->next;
  cdata = (GstCollectData *) collected->data;
  sink2buf = gst_collect_pads_take_buffer (pads, cdata, size);
  
  gst_pad_push(filter->srcpad,sink1buf);
  
  return ret;
  
  goto eos;
 
eos:
  {
    GST_DEBUG_OBJECT (filter, "no data available, must be EOS");
    gst_buffer_unref (outbuf);
    gst_pad_push_event (filter->srcpad, gst_event_new_eos ());
    return -3;
  }
  
}


/* entry point to initialize the plug-in
 * initialize the plug-in itself
 * register the element factories and other features
 */
static gboolean
plugin_init (GstPlugin * plugin)
{
  /* debug category for filtering log messages
   *
   * exchange the string 'Ntoone plugin' with your description
   */
  GST_DEBUG_CATEGORY_INIT (gst_ntoone_debug, "ntoone",
      0, "Template plugin");

  return gst_element_register (plugin, "ntoone", GST_RANK_NONE,
      GST_TYPE_NTOONE);
}

/* PACKAGE: this is usually set by autotools depending on some _INIT macro
 * in configure.ac and then written into and defined in config.h, but we can
 * just set it ourselves here in case someone doesn't use autotools to
 * compile this code. GST_PLUGIN_DEFINE needs PACKAGE to be defined.
 */
#ifndef PACKAGE
#define PACKAGE "pluginntoone"
#endif

/* gstreamer looks for this structure to register plugins
 *
 * exchange the string 'Template plugin' with your plugin description
 */
GST_PLUGIN_DEFINE (
    GST_VERSION_MAJOR,
    GST_VERSION_MINOR,
    "ntoone",
    "Template Example",
    plugin_init,
    "0.10.28",
    "GPL",
    "GStreamer",
    "http://gstreamer.net/"
)

Thanks

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