gmane.comp.video.gstreamer.devel

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

Maxime Louvel — 2012-05-22T15:56:10

HI,

I'm trying to build the following pipeline with a C application :
gst-launch -e udpsrc port=5000 caps="application/x-rtp, media=video,
encoding-name=MP2T-ES" ! rtpmp2tdepay ! mpegtsdemux ! filesink
location=videodstRDO.mpeg

when using gst-launch it works fine, but when I use a C application here is
the error I get (with GST_DEBUG=3) :
0:00:00.032693078 12164 0x1912010 INFO GST_ELEMENT_PADS
gstutils.c:1698:gst_element_link_pads_full: trying to link element
tsdemux:(any) to element mpeg2dec:(any)
0:00:00.032735773 12164 0x1912010 INFO GST_ELEMENT_PADS
gstelement.c:972:gst_element_get_static_pad: no such pad 'video_%04x' in
element "tsdemux"
0:00:00.032759563 12164 0x1912010 INFO GST_ELEMENT_PADS
gstutils.c:1216:gst_element_get_compatible_pad:<tsdemux> Could not find a
compatible pad to link to mpeg2dec:sink

here is the code I wrote :

...
source = gst_element_factory_make ("udpsrc", "rtpsrc");
rtpdepay = gst_element_factory_make ("rtpmp2tdepay", "rtpdepay");
tsdemux = gst_element_factory_make ("mpegtsde

Re: [PATCH 0/2] icydemux doesn't autoplug with master/0.11

Anton Novikov — 2012-05-22T11:38:46

Ok, done; https://bugzilla.gnome.org/show_bug.cgi?id=676549

2012/5/22 Tim-Philipp Müller <t.i.m< at >zen.co.uk>:

Re: [PATCH 0/2] icydemux doesn't autoplug with master/0.11

Tim-Philipp Müller — 2012-05-22T11:20:44

Hi Anton,


thanks for the patches, but could you please put them in bugzilla, so we
don't forget about them?

  http://gstreamer.freedesktop.org/bugs/

  http://gstreamer.freedesktop.org/wiki/SubmittingPatches
  (ignore the outdated bit on how to make patches)

Cheers
 -Tim

[PATCH 2/2] icydemux: activate srcpad before setting caps

Anton Novikov — 2012-05-22T11:10:51

Before gst_pad_set_active() is called, the pad has
FLUSHING flag set, so setting the caps fails
---
 gst/icydemux/gsticydemux.c |    2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/gst/icydemux/gsticydemux.c b/gst/icydemux/gsticydemux.c
index 757d19d..4265529 100644
--- a/gst/icydemux/gsticydemux.c
+++ b/gst/icydemux/gsticydemux.c
< at >< at > -231,6 +231,7 < at >< at > gst_icydemux_add_srcpad (GstICYDemux * icydemux, GstCaps * new_caps)
     g_return_val_if_fail (icydemux->srcpad != NULL, FALSE);
 
     gst_pad_use_fixed_caps (icydemux->srcpad);
+    gst_pad_set_active (icydemux->srcpad, TRUE);
 
     if (icydemux->src_caps) {
       if (!gst_pad_set_caps (icydemux->srcpad, icydemux->src_caps))
< at >< at > -240,7 +241,6 < at >< at > gst_icydemux_add_srcpad (GstICYDemux * icydemux, GstCaps * new_caps)
     GST_DEBUG_OBJECT (icydemux, "Adding src pad with caps %" GST_PTR_FORMAT,
         icydemux->src_caps);
 
-    gst_pad_set_active (icydemux->srcpad, TRUE);
     if (!(gst_element_add_pad (GST_ELEMENT (icydemux), icydemux->srcpa

[PATCH 1/2] icydemux: warning if setting srcpad caps fails

Anton Novikov — 2012-05-22T11:10:50

---
 gst/icydemux/gsticydemux.c |    6 ++++--
 1 file changed, 4 insertions(+), 2 deletions(-)

diff --git a/gst/icydemux/gsticydemux.c b/gst/icydemux/gsticydemux.c
index 4970745..757d19d 100644
--- a/gst/icydemux/gsticydemux.c
+++ b/gst/icydemux/gsticydemux.c
< at >< at > -232,8 +232,10 < at >< at > gst_icydemux_add_srcpad (GstICYDemux * icydemux, GstCaps * new_caps)
 
     gst_pad_use_fixed_caps (icydemux->srcpad);
 
-    if (icydemux->src_caps)
-      gst_pad_set_caps (icydemux->srcpad, icydemux->src_caps);
+    if (icydemux->src_caps) {
+      if (!gst_pad_set_caps (icydemux->srcpad, icydemux->src_caps))
+        GST_WARNING_OBJECT (icydemux, "Failed to set caps on src pad");
+    }
 
     GST_DEBUG_OBJECT (icydemux, "Adding src pad with caps %" GST_PTR_FORMAT,
         icydemux->src_caps);

[PATCH 0/2] icydemux doesn't autoplug with master/0.11

Anton Novikov — 2012-05-22T11:10:49

Hi all,
I've encountered a problem with icydemux not working with decodebin;
patches that fix it for me are following, and can be pulled from my git
tree git://github.com/an146/gst-plugins-good.git; and here's the sample
output with 'warning-if-setting-srcpad-caps-fails' one applied:

$ gst-launch-1.0 --gst-debug=3 playbin uri=http://u16b.di.fm:80/di_ambient  
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
0:00:00.140571669 10141      0x2619b70 WARN                GST_PADS gstpad.c:3432:gst_pad_peer_query:<typefindelement0:src> pad has no peer
0:00:00.768875073 10141      0x2619b70 WARN                GST_PADS gstpad.c:3432:gst_pad_peer_query:<queue2-0:src> pad has no peer
0:00:00.775417927 10141 0x7f0adc0239e0 WARN                icydemux gsticydemux.c:237:gst_icydemux_add_srcpad:<icydemux0> Failed to set caps on src pad
0:00:00.775472401 10141 0x7f0adc0239e0 WARN               decodebin gstdecodebin2.c:1707:analyze_new_pad:<icydemux0:src> pad has ANY caps, not able to autoplug to anything
0:00:0

Re: How to generate error message when there is an internal readerror in one of the object in a pipeline

tanmay.ambre — 2012-05-22T10:53:14

Found the answer.

I need to generate a 

GST_ELEMENT_ERROR messege

which stops the pipeline gracefully.
This is part of my fix.

if (hRes != DS_OK)
{
GST_ERROR("Error Called Called");
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, READ,
        ("Device Error hRes = %i", hRes), (NULL));
GST_DSOUND_UNLOCK (dsoundsrc);
return -1;
}

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rtpjitterbuffer and percent property

Daniel Mellado — 2012-05-22T08:29:33

Hi all!

First of all, thank for your time and knowledge.

I'm using a pipeline wichi has an rtspsrc element on it. Inside this
element, two instances of rtpjitterbuffer are created. I set the buffering
mode in the rtspsrc to Low/High Watermark buffering and then parse the
buffering messages with parse_buffering() to have access to its percentage.

The buffer value goes up to 100% and then the video starts playing. I've
tried a few things, like disconnecting the network, and I can understand
that this value doesn't get updated until it buffers again. But if I ask
directly the jitterbuffers using its percent property, they doesn't show any
change at all. They get stuck at 100% even when they are just empty.

Is this a bug? Am I doing something wrong? Any clues regarding this will be
greatly apretiated.


----Code sample----

self.player=gst.parse_launch(rtspsrc name=source ! decodebin2 !
autovideosink)
bus=self.player.get_bus()
bus.connect("message", self.on_message)
self.source=self.player.get_by_name('sourc

Re: NetBuffer

Tim-Philipp Müller — 2012-05-22T08:25:26


It sounds like you need to *link* to it as well.

Cheers
 -Tim

NetBuffer

enricom — 2012-05-22T08:01:20

Hello, I would like to use the NetBuffer type because I want a user to be
able to send a request to a main program, communicating its ip address. The
problem is that I'm getting the error: " undefined reference to
`gst_netbuffer_new' ".
I already included <gst/netbuffer/gstnetbuffer.h>, do you know if there are
other libraries that need to be included?
Thanks


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Re: Generating frames from a paused video?

Robert Schroll — 2012-05-21T21:18:13

I dismissed this at first, since I'm trying to show the video in only a 
part of an existing Gtk.DrawingArea.  But surely there's a way to make 
clutter render to that DrawingArea instead.  I will investigate more.

Thanks for the suggestion,
Robert

Re: Generating frames from a paused video?

Robert Schroll — 2012-05-21T21:08:39

In a way, I'm glad to hear this - I don't feel so dumb for not figuring 
it out.


If I stick with this solution, I'll check that out.

Thanks for the help,
Robert

How to generate error message when there is an internal read errorin one of the object in a pipeline

tanmay.ambre — 2012-05-21T17:58:51

I am using directsoundsrc plugin. Windows OSSBuild.

There is a gst_directsound_src_read function which has a while loop reading
buffers from the device.

When I accidently unplug the device I found this while loop went into an
infinite loop. I fixed it partially.

I can detect when there is error in capture.

What I have not figured out is now how to make the pipeline stop.

while (length > dwBufferSize) {
    Sleep (dsoundsrc->latency_time / 1000);

    hRes =
        IDirectSoundCaptureBuffer_GetCurrentPosition
(dsoundsrc->pDSBSecondary,
        &dwCurrentCaptureCursor, NULL);
if (hRes != 0)
{
GST_ERROR("\nDevice Error HRESULT = %i",hRes);///This is where i can
detect an error
break;
}

    /* calculate the buffer */
    if (dwCurrentCaptureCursor < dsoundsrc->current_circular_offset) {
      dwBufferSize = dsoundsrc->buffer_size -
          (dsoundsrc->current_circular_offset - dwCurrentCaptureCursor);
    } else {
      dwBufferSize =
          dwCurrentCaptureCursor - dsoundsrc->current_circ

Re: Problems with segment seek on audiotestsrc

Simon Berg — 2012-05-21T17:30:15

I will.

Unfortunately that doesn't seem to work. I converted the sample index to
GstClockTime and added 1ns to make sure it would convert back to the
correct number of sample. The problem was that the last buffer returned
to GstBaseSrc from the create method of the subclass had a timestamp
that was smaller than the segment end so it was called an extra time
which resulted in a full buffer. 
How is a subclass of GstBaseSrc supposed to behave when the requested
length doesn't match a whole number of samples (or frames)?
Especially if the time is just slightly passed a full sample.

There are probably even more problems when mixing times and samples,
but I have to think that through a bit more.

simon

Re: Need clarification on gstcolorbalance

Yogesh Marwaha — 2012-05-21T14:46:26

Now I get the following:-
Channel name: XV_BRIGHTNESS
Channel name: XV_CONTRAST

I was also expecting HUE and SATURATION to be listed.

Anyways, my main problem is that values of these properties provided
by xvimagesink & videobalance elements are imcompatiable, so I was
searching for a standard way of setting their values so that I may use
same code, no matter I'm using xvimagesink or videobalance+ximagesink.

What do you recommend? Would I have to/(Should I) use separate code
for both cases? Or do I forget that xvimagesink offers these
properties?

PS: Upto now I had been thinking that standard property names in
gstreamer (e.g. the volume property) imply standard values, and all my
code also assumes that.

Regards,

Yogesh Marwaha
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Re: Can I have two sinks using the same chain function?

iron_guitarist1987 — 2012-05-21T14:34:18

Could I then have them as request pads, explicitly create only one video sink
pad, and one text sink pad, and them put them in a collection? I was trying
to use the gstadder, but it has so many functions which I'm not sure they
are necessary or not, like: 

gst_adder_dispose
gst_adder_query
gst_adder_src_event
gst_adder_sink_event
gst_adder_request_new_pad
gst_adder_release_pad
gst_adder_change_state 
gst_adder_do_clip
gst_adder_collected 

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Re: Can I have two sinks using the same chain function?

iron_guitarist1987 — 2012-05-21T14:27:20

I am using the incoming text data to do some image processing on the video
(the text is not subtitles). I was looking into the Cairo text overlay, but
what I understood is that it takes the text from a property, and I really
didn't understand why it has to sink pads.

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pipeline doesn't work

Jones — 2012-05-21T09:36:57

Hi,
I'm quite new to geastreamer and i have problem with build working pipeline.
After launch that pipe i doesn't get any errors, just window is open and
it's freezing.

Thanks for any suggestions.

pipeline = gst_pipeline_new("pipeline");
src = gst_element_factory_make ("filesrc", "mysrc");
g_object_set(src, "location", "D:\\file.avi", NULL);

    queue= gst_element_factory_make("queue", "vqueue");
queue2 = gst_element_factory_make("queue", "queue1");
    flt = gst_element_factory_make("capsfilter", "vfilter");
    conv = gst_element_factory_make("ffmpegcolorspace", "vconv");
    scale= gst_element_factory_make("videoscale", "vscale");
    vsink= gst_element_factory_make("dshowvideosink", "vsink");
decoder = gst_element_factory_make("ffdec_msmpeg4v2", "decoder");
mp3parse = gst_element_factory_make("mp3parse", "parser");
ffdec_mp3 = gst_element_factory_make("ffdec_mp3", "decodermp3");
wavenc = gst_element_factory_make("wavenc", "wavenc");
wavparse = gst_element_factory_make("wavparse", "wavparse");

Re: Generating frames from a paused video?

Tim-Philipp Müller — 2012-05-21T08:54:52

Hi,


I think it would be best/easiest if you did the overlay outside of the
GStreamer pipeline. Have you considered using e.g. clutter and
clutter-gst?

Cheers
 -Tim

Re: Video synchronization

enricom — 2012-05-21T08:43:34

Well... I already tried many times but I failed to receive a precise video
position. In my opinion that code returns the position of the video player,
so if it's receiving frames at constant rate it's good (it works if the file
I'm playing is on my hard disk). 
But in my case I have a rtp source, and it happens that the video player
shows the first frame, then there's a delay (so the first frame freezes) and
after that it goes on displaying the successive frames correctly. But I
found that during that delay the timer of the video player goes on...so for
example if we have timestamp = 0 shown at video player position = 0 , then
there's that delay and timestamp 0.04 is shown at video player position =
0.8 (it's only an example)...and so I'm losing reference between my
timestamp and the video player position...

Anyway by querying the video player position (with a DEFAULT format) and
checking buffers passing the identity just before the player I'm able to
correctly identify  the timestamp that is shown (of cour

Re: Udp streaming

enricom — 2012-05-21T08:32:49

Ok, found, memcpy fits my needs...

Bye!



enricom wrote


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