gstreamer-embedded — GStreamer on embedded platforms

Re: [gst-embedded] is jrtplib required for RTCP reports?

[Felipe C. <fel...@gm...>]

Wrong mailing list, this is the correct one:

gst...@li...

-- 
Felipe Contreras

[gst-embedded] is jrtplib required for RTCP reports?

[ranjit r. <ran...@ya...>]

Dear All,
I want to get the RTCP reports.I am using the following pipeline:  udpsrc ! gstrtpbin ! rtpdepay ! decoder ! display.
I am setting udpsrc's port property as device port + 1.linked udpsrc's "src" pad with "recv_rtcp_sink_1" pad of gstrtpbin.tapped a buffer probe on src pad of udpsrc.
my question is : in the buffer probe call back,        will i get the rtcp reports on the src pad of udpsrc.        in what form will these reports be?        i am not using jrtplib explicilty, i read in the docs that jrtplib is required, do i need it for collecting RTCP reports? if so how to use it?
please clarify my above queries.Thanks in advance,Ranjit.


      

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[Tim-Philipp M. <t....@ze...>]

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[gst-embedded] gstreamer for android

[B U J J I <siv...@gm...>]

Hi all,

I have seen that gstreamer-0.10.32 has support for android.

but i didnt get how to build it for android..

can anybody help how to configure and build it for android..

Thanks,
Bujji

Re: [gst-embedded] Syncing in TIDmaiVideoSink

[Andreas A. <a....@zy...>]

Thanks for your reply. Do you have a tutorial which shows how I can get 
gst-dsp running on my IGEPv2 board?? I already have compiled the sources 
from "git://github.com/felipec/gst-dsp.git".

Do I need to load the bridgedriver kernel module to use gst-dsp?? Which 
kernel version do you use? Is it one of the kernels which are available 
at the GIT repository of IGEP??

Thanks,
Andreas

On 2011-02-01 23:55, Sjoerd Simons wrote:
> On Mon, 2011-01-31 at 23:33 +0200, Marco Ballesio wrote:
>> yes, if the upstream elements declare a proper latency. It's possibly
>> a bug for the decoder element, not maintained by this community
>> afaik.. have you considered / tried using gst-dsp instead (I don't
>> know whether it runs on your particular hw)?
>
> Fwiw, i've used gst-dsp on my IGEPv2 boards and it works quite well and
> doesn't seem to have any of the latency issue that people seem to
> encounter with the TI elements.
>

Re: [gst-embedded] Syncing in TIDmaiVideoSink

[Sjoerd S. <sj...@lu...>]

On Mon, 2011-01-31 at 23:33 +0200, Marco Ballesio wrote:
> yes, if the upstream elements declare a proper latency. It's possibly
> a bug for the decoder element, not maintained by this community
> afaik.. have you considered / tried using gst-dsp instead (I don't
> know whether it runs on your particular hw)?

Fwiw, i've used gst-dsp on my IGEPv2 boards and it works quite well and
doesn't seem to have any of the latency issue that people seem to
encounter with the TI elements.

-- 
Sjoerd Simons <sj...@lu...>

Re: [gst-embedded] Syncing in TIDmaiVideoSink

[Andreas A. <a....@zy...>]

Thank you for your reply and the interesting links. I wonder why I 
haven't found them with google.
BTW the decoder always buffers 1.5MB of data before decoding. But in the 
case you are getting the data from a file it cannot be noticed that the 
buffer is that big. I'll try to change this behavior after I have solved 
the synchronization issue.

On 2011-01-31 22:33, Marco Ballesio wrote:
> In short, it is the decoder which uses -under some conditions- a
> fixed-size buffer for storing encoded data, this meaning that, in case
> of low resolutions, frame rates and -especially- bitrates it will take
> a few seconds before the video decoder emits anything. I just wonder
> if/why the element is not properly declaring its latency.

You are right the video decoder doesn't declare its latency in a correct 
way. I found it yesterday while debugging and looking at the code. I 
think if the latency query is correct the large buffer shouldn't be a 
problem.

> Said so, a few workarounds are possible, the easiest one being an
> increase in the bitrate feeding the decoder (not possible in all the
> cases though).

Increasing the bitrate isn't possible for me. I think the only way for 
me is to declare the correct latency. Curious that this haven't been 
done by TI itself.

Thanks for your help,
Andreas

-- 
DI Andreas Auer                     aauer1 (at) gmail.com
http://about.me/Andreas.Auer

Re: [gst-embedded] Syncing in TIDmaiVideoSink

[Marco B. <gib...@gm...>]

Hi,

this question is periodically coming to GStreamer mailing lists (I
guess it depends on the Moon phases ;) ).

I must admit I never had access to any hw using TIDmaiVideoSink but it
appears I've got lots of experience on the subject :D.

Some pointers:

http://gstreamer-devel.966125.n4.nabble.com/Video-and-Audio-sync-problem-td2322769.html

http://gstreamer-devel.966125.n4.nabble.com/long-pauses-during-rtsp-playback-td2997863.html

http://gstreamer-devel.966125.n4.nabble.com/long-pauses-when-viewing-RTSP-stream-td3001831.html

http://gstreamer-devel.966125.n4.nabble.com/Help-on-execution-speed-and-optimization-in-gst-launch-td3005694.html

... and this (which is suspiciously similar to your issue)

http://gstreamer-devel.966125.n4.nabble.com/Help-on-execution-speed-and-optimization-in-gst-launch-td3005694.html

In short, it is the decoder which uses -under some conditions- a
fixed-size buffer for storing encoded data, this meaning that, in case
of low resolutions, frame rates and -especially- bitrates it will take
a few seconds before the video decoder emits anything. I just wonder
if/why the element is not properly declaring its latency.

Said so, a few workarounds are possible, the easiest one being an
increase in the bitrate feeding the decoder (not possible in all the
cases though).

On Mon, Jan 31, 2011 at 10:28 AM, Andreas Auer <a....@zy...> wrote:
..snip..
> Does anybody know if there is a patch for this problem? Why does the
> GstBaseSink believe the frames are too late. IMHO the GstBaseSink
> shouldn't take the latency of the upstream elements into account.

yes, if the upstream elements declare a proper latency. It's possibly
a bug for the decoder element, not maintained by this community
afaik.. have you considered / tried using gst-dsp instead (I don't
know whether it runs on your particular hw)?

Regards

>
> Cheers,
> Andreas
>
> --
> DI Andreas Auer                     aauer1 (at) gmail.com
> http://about.me/Andreas.Auer
>
>
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>

[gst-embedded] Syncing in TIDmaiVideoSink

[Andreas A. <a....@zy...>]

Hello,

I hope someone can help me. I have an IGEPv2 board with an OMAP3530 cpu. 
Currently, I'm trying to stream an H264 rtp stream to the board decode 
it on the board with the DSP and show the decoded video on the screen.
The problem is that the video output with the TIDmaiVideoSink does not 
sync correctly. The problem is that the decoder buffers quite long 
(about 30 seconds). So, the first decoded frame is displayed with a 
delay of 30 seconds. And therefore the base class "GstBaseSink" wants to 
regain the past 30 seconds and plays the decoded frames as fast es 
possible.

Does anybody know if there is a patch for this problem? Why does the 
GstBaseSink believe the frames are too late. IMHO the GstBaseSink 
shouldn't take the latency of the upstream elements into account.

Cheers,
Andreas

-- 
DI Andreas Auer                     aauer1 (at) gmail.com
http://about.me/Andreas.Auer

Re: [gst-embedded] Problem : Gstreamer on android by NDK

[Changqing W. <wei...@gm...>]

Could anybody share steps to build gstreamer on android, I am using ieei's
git branch(http://github.com/ieei). The glib repository is build just using
ndk-build and it works smoothly,generate gobject, gmodule, gthread. However
I have no idea how to build gstreamer.

For glib, I simple create a jni path for all git clones and ndk-build, and
the build finished without any error. But gstreamer should have depenencies
on glib, I have no idea how to add this dependencies.

Anybody could provide any help?

[gst-embedded] amr-wb seek problem

[Jorney <jor...@as...>]

I am writing amr-wb player with gstreamer on Marvell pxa310.
When I seek to a position by time format, there is a long time blocking
(10 sec or even to 1 min), and there is no voice during the block, then
an EOS message was emitted,
but the position I seeked is far away to the end of the file.

Bytes format has the same problem! here is my seek code:

breslt = gst_element_seek(pipeline,
1.0,
GST_FORMAT_BYTES,
GST_SEEK_FLAG_FLUSH,//|GST_SEEK_FLAG_KEY_UNIT|GST_SEEK_FLAG_SEGMENT,
GST_SEEK_TYPE_SET,
bytePos,//GST_SECOND*nForw*count,
GST_SEEK_TYPE_NONE,
GST_CLOCK_TIME_NONE
);

I use gstreamer-0.10.29 and gst-plugins-ugly-0.10.15's amrwbdec,
gst-plugins-bad-0.10.13's amrparse. like this:

gst-launch filesrc location=abc.amr ! amrparse ! amrwbdec ! alsasink

Could any body help?




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[gst-embedded] [Fwd: amr-wb seek problem]

[Jorney <jor...@as...>]

And the codec is opencore-amr-0.1.2


-------- 原始信息 --------
主题: 	amr-wb seek problem
日期: 	Sat, 08 Jan 2011 16:54:28 +0800
发件人: 	Jorney <jor...@as...>
收件人: 	gst...@li...



I am writing amr-wb player with gstreamer on Marvell pxa310.
When I seek to a position by time format, there is a long time blocking
(10 sec or even to 1 min), and there is no voice during the block, then
an EOS message was emitted,
but the position I seeked is far away to the end of the file.

Bytes format has the same problem! here is my seek code:

breslt = gst_element_seek(pipeline,
1.0,
GST_FORMAT_BYTES,
GST_SEEK_FLAG_FLUSH,//|GST_SEEK_FLAG_KEY_UNIT|GST_SEEK_FLAG_SEGMENT,
GST_SEEK_TYPE_SET,
bytePos,//GST_SECOND*nForw*count,
GST_SEEK_TYPE_NONE,
GST_CLOCK_TIME_NONE
);

I use gstreamer-0.10.29 and gst-plugins-ugly-0.10.15's amrwbdec,
gst-plugins-bad-0.10.13's amrparse. like this:

gst-launch filesrc location=abc.amr ! amrparse ! amrwbdec ! alsasink

Could any body help?






=====================================================================================================================================
This email and any attachments to it contain confidential information and are intended solely for the use of the individual to whom it 
is addressed.If you are not the intended recipient or receive it accidentally, please immediately notify the sender by e-mail and delete 
the message and any attachments from your computer system, and destroy all hard copies. If any, please be advised that any unauthorized 
disclosure, copying, distribution or any action taken or omitted in reliance on this, is illegal and prohibited. Furthermore, any views 
or opinions expressed are solely those of the author and do not represent those of ASUSTeK. Thank you for your cooperation.
=====================================================================================================================================

[gst-embedded] (no subject)

[Jesse B. <Je...@sp...>]

I am having some problems getting the dfbvideosink to play a video file. I
was able to use the example provided and play from the test video source.
After some searching I decided to try the playbin but I don't get any video
from that. How should I set up the pipeline? Here is what I have so far

 

 

#include <directfb.h>

#include <gst/gst.h>

 

static IDirectFB *dfb = NULL;

static IDirectFBSurface *primary = NULL;

static GMainLoop *loop;

 

#define DFBCHECK(x...)                                         \

  {                                                            \

    DFBResult err = x;                                         \

                                                               \

    if (err != DFB_OK)                                         \

      {                                                        \

        fprintf( stderr, "%s <%d>:\n\t", __FILE__, __LINE__ ); \

        DirectFBErrorFatal( #x, err );                         \

      }                                                        \

  }

 

static gboolean

get_me_out (gpointer data)

{

  g_main_loop_quit (loop);

  return FALSE;

}

 

int

main (int argc, char *argv[])

{

  DFBSurfaceDescription dsc;

  GstElement *pipeline, *sink;

 

  /* Init both GStreamer and DirectFB */

  DFBCHECK (DirectFBInit (&argc, &argv));

  gst_init (&argc, &argv);

 

  /* Creates DirectFB main context and set it to fullscreen layout */

  DFBCHECK (DirectFBCreate (&dfb));

  DFBCHECK (dfb->SetCooperativeLevel (dfb, DFSCL_FULLSCREEN));

 

  /* We want a double buffered primary surface */

  dsc.flags = DSDESC_CAPS;

  dsc.caps = DSCAPS_PRIMARY | DSCAPS_FLIPPING;

 

  DFBCHECK (dfb->CreateSurface (dfb, &dsc, &primary));

 

  /* Creating our pipeline : videotestsrc ! dfbvideosink */

  pipeline = gst_element_factory_make("playbin", NULL); //gst_pipeline_new
(NULL);

  g_assert (pipeline);

 

  sink = gst_element_factory_make ("dfbvideosink", NULL);

  g_assert (sink);

  /* That's the interesting part, giving the primary surface to dfbvideosink
*/

  g_object_set (sink, "surface", primary, NULL);

 

  g_object_set(pipeline, "uri", "file:///root/test.mp4
<file:///\\root\test.mp4> ", NULL);

  g_object_set(pipeline, "video-sink", sink, NULL);

 

  /* Let's play ! */

  gst_element_set_state (pipeline, GST_STATE_PLAYING);

 

  /* we need to run a GLib main loop to get out of here */

  loop = g_main_loop_new (NULL, FALSE);

  /* Get us out after 20 seconds */

  g_timeout_add (20000, get_me_out, NULL);

  g_main_loop_run (loop);

 

  /* Release elements and stop playback */

  gst_element_set_state (pipeline, GST_STATE_NULL);

 

  /* Free the main loop */

  g_main_loop_unref (loop);

 

  /* Release DirectFB context and surface */

  primary->Release (primary);

  dfb->Release (dfb);

 

  return 0;

}

 

The result is a brown screen (the default directfb color I think).

 

Thanks for your help

-Jesse

 

 

[gst-embedded] How to use dfbvideosink

[Jesse B. <Je...@sp...>]

I am having some problems getting the dfbvideosink to play a video file. I
was able to use the example provided and play from the test video source.
After some searching I decided to try the playbin but I don't get any video
from that. How should I set up the pipeline? Here is what I have so far

 

 

#include <directfb.h>

#include <gst/gst.h>

 

static IDirectFB *dfb = NULL;

static IDirectFBSurface *primary = NULL;

static GMainLoop *loop;

 

#define DFBCHECK(x...)                                         \

  {                                                            \

    DFBResult err = x;                                         \

                                                               \

    if (err != DFB_OK)                                         \

      {                                                        \

        fprintf( stderr, "%s <%d>:\n\t", __FILE__, __LINE__ ); \

        DirectFBErrorFatal( #x, err );                         \

      }                                                        \

  }

 

static gboolean

get_me_out (gpointer data)

{

  g_main_loop_quit (loop);

  return FALSE;

}

 

int

main (int argc, char *argv[])

{

  DFBSurfaceDescription dsc;

  GstElement *pipeline, *sink;

 

  /* Init both GStreamer and DirectFB */

  DFBCHECK (DirectFBInit (&argc, &argv));

  gst_init (&argc, &argv);

 

  /* Creates DirectFB main context and set it to fullscreen layout */

  DFBCHECK (DirectFBCreate (&dfb));

  DFBCHECK (dfb->SetCooperativeLevel (dfb, DFSCL_FULLSCREEN));

 

  /* We want a double buffered primary surface */

  dsc.flags = DSDESC_CAPS;

  dsc.caps = DSCAPS_PRIMARY | DSCAPS_FLIPPING;

 

  DFBCHECK (dfb->CreateSurface (dfb, &dsc, &primary));

 

  /* Creating our pipeline : videotestsrc ! dfbvideosink */

  pipeline = gst_element_factory_make("playbin", NULL); //gst_pipeline_new
(NULL);

  g_assert (pipeline);

 

  sink = gst_element_factory_make ("dfbvideosink", NULL);

  g_assert (sink);

  /* That's the interesting part, giving the primary surface to dfbvideosink
*/

  g_object_set (sink, "surface", primary, NULL);

 

  g_object_set(pipeline, "uri", "file:///root/test.mp4", NULL);

  g_object_set(pipeline, "video-sink", sink, NULL);

 

  /* Let's play ! */

  gst_element_set_state (pipeline, GST_STATE_PLAYING);

 

  /* we need to run a GLib main loop to get out of here */

  loop = g_main_loop_new (NULL, FALSE);

  /* Get us out after 20 seconds */

  g_timeout_add (20000, get_me_out, NULL);

  g_main_loop_run (loop);

 

  /* Release elements and stop playback */

  gst_element_set_state (pipeline, GST_STATE_NULL);

 

  /* Free the main loop */

  g_main_loop_unref (loop);

 

  /* Release DirectFB context and surface */

  primary->Release (primary);

  dfb->Release (dfb);

 

  return 0;

}

 

The result is a brown screen (the default directfb color I think).

 

Thanks for your help

-Jesse

 

Re: [gst-embedded] rtsp client with rtpdec problem

[Marco B. <gib...@gm...>]

Ok, looks like for the n-th time I just replied to the sender instead
of the mailing list..

2010/12/19 Marco Ballesio <gib...@gm...>:
> Hi,
>
> On Sat, Dec 4, 2010 at 10:59 AM,  <Alb...@it...> wrote:
>> Dear,
>>
>> I design a rtsp client player using gst-plugin-good,
>>
>> but I using "rtpdec" replace "rtpbin",
>>
>> my client could receive "sender report" from server,
>>
>> but could not send "receive report" to the server.
>>
>> I try using "rtpbin" , and it could do both receive/send  actions.
>>
>> How could I send "receive report" to the server when I using "rtpdec" ?
>>
>
> similarly to what happens in gstrtpbin, you must use the same value
> for <id> when connecting the pads rtcp_src_<id> and
> recv_rtp_src_<id>_<ssrc>_<pt>. This way the gstrtpsession element
> within the rtpdec will send RTCP packets (likely one each 5 seconds)
> to the udpsink where you connected rtcp_src_<id>.
>
> Regards
>
>> regards,
>>
>> Albert
>>
>> 本信件可能包含工研院機密資訊，非指定之收件者，請勿使用或揭露本信件內容，並請銷毀此信件。
>> This email may contain confidential information. Please do not use or
>> disclose it in any way and delete it if you are not the intended recipient.
>> ------------------------------------------------------------------------------
>>
>> _______________________________________________
>> Gstreamer-embedded mailing list
>> Gst...@li...
>> https://lists.sourceforge.net/lists/listinfo/gstreamer-embedded
>>
>>
>

[gst-embedded] Error : Linking exe with gstreamer shared lib

[amit s. <ami...@gm...>]

Hi,

I have successfully created lib for gstreamer and supported plugin by
using NDK, now i want to test the working of so file, for that I
create  a application and create a exe by using NDK .

Next step is to run the exe on android-emulator, for that first I load
the exe into directory /data/temp by using adb push command, then push
all the shared library which is required to execute the exe i.e.

libglib-2.0.so
libgmodule-2.0.so
libgobject-2.0.so
libgthread-2.0.so
libgstcoreelements-gstplugin.so
libgstreamer-0.10-libs.so
libgstcoreindexers-gstplugin.so
libgstreamer.so
libgst-plugins-base-libs-0.10.so
libgstapp.so

So everything is now in the same folder.

Next step is to link the library at run time so for that I set the
path by using LD_LIBRARY_PATH

So up to now everything seem to be OK.

Final Step: To execute the exe.

Error: The strange thing that i observe that exe is able to execute
some of the basic api of gstreamer like as,

gst_init (&argc, &argv);
bin = gst_pipeline_new ("pipeline");

But while executing following command, exe not able to create appsrc
element.

appsrc = gst_element_factory_make ("appsrc", NULL);    --------> Not
able to create this element

I have crossed check and I found I have all the .so file that is
required to execute the code.

GLib: Cannot convert message: Could not open converter from 'UTF-8' to
'ASCII' ----> I am getting this message also while executing the code.

So, If any body face such type of error or want to share something on
this issue then please share your valueable point.

Thanks

Re: [gst-embedded] Playback of raw audio on N900: Internal data flow error

[Thomas J. <to...@si...>]

Am Freitag, 17. Dezember 2010, 00:51:55 schrieb Tim-Philipp Müller:
>   audio_caps = gst_caps_new_simple("audio/x-raw-int",
>       "width", G_TYPE_INT, (gint)16,
>       "depth", G_TYPE_INT, (gint)16,
>       "channels" ,G_TYPE_INT, (gint)2,
>  -->  "signed",G_TYPE_INT,1, <--
>       "rate", G_TYPE_INT, 44100,
>       "endianness", G_TYPE_INT, (gint)1234,
>       NULL);
> 
> The signed field should be of G_TYPE_BOOLEAN.

I was surprised to hear the audio when I changed it to bool :o) Nice!

Also it looks like I can push arbitrary data sizes into the appsrc without the 
audioparse element. It's working for 10 minutes like this... *fingers crossed*

Thanks again,
Thomas

Re: [gst-embedded] Unable to stream MPEG4 or H.264 from BeagleBoard

[Matthew B. <mj...@le...>]

Andrey,

Thank you very much for your suggestions and I'll give them a try!

--
Matthew Braun
mj...@le...





On 12/16/10 6:52 AM, "Andrey Nechypurenko" <and...@ya...> wrote:

>Hi Matthew,
>
>>gst-launch -v v4l2src ! TIVidenc1 codecName=h264enc
>>engineName=codecServer ! rtph264pay pt=96 ! udpsink host=<DEST> port=5000
>
>I think there is missing colorconversion element (ffmpegcolorspace)
>before 
>TIVidenc1. Also, it is necessary to specify colorspace (fourcc)
>explicitly. You 
>can take a look at this thread which discuss similar issue:
>
>https://gstreamer.ti.com/gf/project/gstreamer_ti/forum/?_forum_action=Foru
>mMessageBrowse&thread_id=3251&action=ForumBrowse&forum_id=187
>
>
>In addition you can take a look at some example pipelines I am using in
>my 
>project:
>http://www.gitorious.org/veter/veter/blobs/master/misc/car.config#line36
>
>HTH,
>Andrey.
>
>----- Original Message ----
>From: Matthew Braun <mj...@le...>
>To: gst...@li...
>Sent: Thu, December 9, 2010 5:29:15 PM
>Subject: [gst-embedded] Unable to stream MPEG4 or H.264 from BeagleBoard
>
>Greetings, and apologies in advance if my question is excessively simple,
>although my searches have so far proven futile.
>
>I'm running Angstrom on a BeagleBoard C (OMAP3530) with the gstreamer-ti
>package (GStreamer 0.10.30) installed. At boot the following commands are
>run:
>
>cd /usr/share/ti/gst/omap3530/
>./loadmodules.sh
>export GST_REGISTRY=/tmp/gst_registry.bin
>export LD_LIBRARY_PATH=/usr/lib
>export GST_PLUGIN_PATH=/usr/lib/gstreamer-0.10
>export PATH=/usr/bin:$PATH
>cat /dev/zero > /dev/fb2 2> /dev/null
>
>
>Running the command
>
>gst-launch v4l2src ! video/x-raw-yuv,width=640,height=480 !
>ffmpegcolorspace ! jpegenc ! multipartmux ! udpsink host=<DEST> port=5000
>
>Succeeds (I can open VLC on <DEST> and view the stream, although it's
>pretty choppy). However, I'd like to do an MPEG4 stream instead. Following
>the example from
>http://processors.wiki.ti.com/index.php/Example_GStreamer_Pipelines#OMAP35
>x
>, I tried:
>
>gst-launch-0.10  v4l2src ! 'video/x-raw-yuv,width=640,height=480' !
>ffenc_mpeg4 ! rtpmp4vpay ! udpsink host=169.254.12.28 port=5000 -v
>
>But I get
>
>Setting pipeline to PAUSED ...
>/GstPipeline:pipeline0/GstV4l2Src:v4l2src0.GstPad:src: caps =
>video/x-raw-yuv, format=(fourcc)I420, width=(int)640, height=(int)480,
>framerate=(fraction)30/1
>Pipeline is live and does not need PREROLL ...
>Setting pipeline to PLAYING ...
>New clock: GstSystemClock
>/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps =
>video/x-raw-yuv, format=(fourcc)I420, width=(int)640, height=(int)480,
>framerate=(fraction)30/1
>/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:sink: caps =
>video/x-raw-yuv, format=(fourcc)I420, width=(int)640, height=(int)480,
>framerate=(fraction)30/1
>/GstPipeline:pipeline0/ffenc_mpeg4:ffenc_mpeg40.GstPad:src: caps =
>video/mpeg, width=(int)640, height=(int)480, framerate=(fraction)30/1,
>mpegversion=(int)4, systemstream=(boolean)false
>/GstPipeline:pipeline0/ffenc_mpeg4:ffenc_mpeg40.GstPad:sink: caps =
>video/x-raw-yuv, format=(fourcc)I420, width=(int)640, height=(int)480,
>framerate=(fraction)30/1
>/GstPipeline:pipeline0/GstRtpMP4VPay:rtpmp4vpay0.GstPad:sink: caps =
>video/mpeg, width=(int)640, height=(int)480, framerate=(fraction)30/1,
>mpegversion=(int)4, systemstream=(boolean)false
>/GstPipeline:pipeline0/GstRtpMP4VPay:rtpmp4vpay0.GstPad:src: caps =
>application/x-rtp, media=(string)video, clock-rate=(int)90000,
>encoding-name=(string)MP4V-ES, profile-level-id=(string)1,
>config=(string)000001b001000001b58913000001000000012000c48d8800f514043c146
>3
>000001b24c61766335322e39372e32, payload=(int)96, ssrc=(uint)1464948709,
>clock-base=(uint)3097574894, seqnum-base=(uint)47019
>/GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps =
>application/x-rtp, media=(string)video, clock-rate=(int)90000,
>encoding-name=(string)MP4V-ES, profile-level-id=(string)1,
>config=(string)000001b001000001b58913000001000000012000c48d8800f514043c146
>3
>000001b24c61766335322e39372e32, payload=(int)96, ssrc=(uint)1464948709,
>clock-base=(uint)3097574894, seqnum-base=(uint)47019
>libv4l2: error converting / decoding frame data: v4l-convert: error
>parsing JPEG header: Not a JPG file ?
>Caught SIGSEGV accessing address 0x4
>unable to fork gdb: Cannot allocate memory
>Spinning.  Please run 'gdb gst-launch 1582' to continue debugging, Ctrl-C
>to quit, or Ctrl-\ to dump core.
>
>
>
>I assume this is related to the error parsing the JPEG header? The v4l2
>source is a Logitech C310 USB webcam.
>
>Trying to encode as h.264 instead, from the same examples, I use:
>gst-launch -v v4l2src ! TIVidenc1 codecName=h264enc
>engineName=codecServer ! rtph264pay pt=96 ! udpsink host=<DEST> port=5000
>-v
>But I get:
>Setting pipeline to PAUSED ...
>ERROR: Pipeline doesn't want to pause.
>ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: Could not
>negotiate format
>Additional debug info:
>  gstbasesrc.c(2755): gst_base_src_start ():
>/GstPipeline:pipeline0/GstV4l2Src:v4l2src0:
>Check your filtered caps, if any
>Setting pipeline to NULL ...
>Freeing pipeline ...
>
>
>Cat /proc/cmdline gives:
>console=ttyS2,115200n8 console=tty0 root=/dev/mmcblk0p2 rw rootwait
>rootdelay=2 mem=80M video=omapfb:vram:2M,vram:4M
>
>
>
>Any suggestions as to what I might be missing or something I should be
>trying instead to get MPEG4 or H.264 streaming to work?
>
>Thank you very much for any guidance or suggestions!
>
>
>--
>Matthew Braun
>mj...@le...
>
>
>
>
>
>--------------------------------------------------------------------------
>----
>_______________________________________________
>Gstreamer-embedded mailing list
>Gst...@li...
>https://lists.sourceforge.net/lists/listinfo/gstreamer-embedded
>
>
>      

Re: [gst-embedded] Playback of raw audio on N900: Internal data flow error

[Tim-Philipp M. <t....@ze...>]

On Fri, 2010-12-17 at 09:32 +0100, Thomas Jarosch wrote:

> > What's the debug message detail on that? (not-negotiated?)
>
> I tried to enable debug output via the environment variable and it didn't 
> work. So I suspected it's not enabled during compile time on the N900. I'll 
> retry with the "--gst-debug-level" command line switch.

Debug output would be nice indeed, but I didn't expect that to be
enabled by default on the N900. What I meant was the debug message
detail of the error message (an error message contains two things
usually: an error to display to the user, and a second debug string with
more details for debugging purposes).


> Ok. One more silly question about the caps. For audio channels,
> there's a property called "width" and "depth". Here's an example
> gst-inspect output of a alsasink:
> 
>       audio/x-raw-int
>              endianness: { 1234, 4321 }
>                  signed: { true, false }
>                   width: 32
>                   depth: 24
>                    rate: [ 1, 2147483647 ]
>                channels: [ 1, 2147483647 ]
> 
> Does it mean it supports 24-bit samples
> stored as a 32bit integers?

Yes.

> > > Can I push arbitrary data size into the appsrc or must it be
> > > in sample size chunks?
> > 
> > That depends on what you're pushing into appsrc. If it's raw PCM audio,
> > it should be in chunks of bytes_per_sample * channels.
> 
> This might be the issue, I just push anything I get from rockbox in there.
> I hoped the gstreamer framework would handle the reassembling for me.

You can insert an audioparse element after appsrc, then it will chunk
things for you (if you set the right format via the properties). It will
also set caps for you then, so you don't have to do that with appsrc.

Cheers
 -Tim

Re: [gst-embedded] Playback of raw audio on N900: Internal data flow error

[Thomas J. <to...@si...>]

Hi Tim,

On Friday, 17. December 2010 00:51:55 Tim-Philipp Müller wrote:
> On Fri, 2010-12-17 at 00:10 +0100, Thomas Jarosch wrote:
> > I tried all kind of ways to push the raw audio via an "appsrc" to
> > "playbin2". The result is always an "Internal data flow" error.
> 
> What's the debug message detail on that? (not-negotiated?)

Thanks for your reply.

I tried to enable debug output via the environment variable and it didn't 
work. So I suspected it's not enabled during compile time on the N900. I'll 
retry with the "--gst-debug-level" command line switch.

> > Please see the attached code and the function call log.
> > The magic happens in pcm_play_dma_init(), found_source() and
> > feed_data().
> > 
> > Any idea what could be wrong? The timestamps on the GstBuffer?
> 
> That depends on the exact error, but if it's not-negotiated it's likely
> got something to do with caps.

Ok. One more silly question about the caps. For audio channels,
there's a property called "width" and "depth". Here's an example
gst-inspect output of a alsasink:

      audio/x-raw-int
             endianness: { 1234, 4321 }
                 signed: { true, false }
                  width: 32
                  depth: 24
                   rate: [ 1, 2147483647 ]
               channels: [ 1, 2147483647 ]

Does it mean it supports 24-bit samples
stored as a 32bit integers?

> > Can I push arbitrary data size into the appsrc or must it be
> > in sample size chunks?
> 
> That depends on what you're pushing into appsrc. If it's raw PCM audio,
> it should be in chunks of bytes_per_sample * channels.

This might be the issue, I just push anything I get from rockbox in there.
I hoped the gstreamer framework would handle the reassembling for me.

> > The code is quite hacky at the moment as I want to get
> > it up and running and then beautify it. Sorry:)
> 
>   audio_caps = gst_caps_new_simple("audio/x-raw-int",
>       "width", G_TYPE_INT, (gint)16,
>       "depth", G_TYPE_INT, (gint)16,
>       "channels" ,G_TYPE_INT, (gint)2,
>  -->  "signed",G_TYPE_INT,1, <--
>       "rate", G_TYPE_INT, 44100,
>       "endianness", G_TYPE_INT, (gint)1234,
>       NULL);
> 
> The signed field should be of G_TYPE_BOOLEAN.

Thanks! The error must have happened when I added/tweaked
various settings to find the source of the error.

Cheers,
Thomas

Re: [gst-embedded] Playback of raw audio on N900: Internal data flow error

[Tim-Philipp M. <t....@ze...>]

On Fri, 2010-12-17 at 00:10 +0100, Thomas Jarosch wrote:

> I tried all kind of ways to push the raw audio via an "appsrc" to "playbin2".
> The result is always an "Internal data flow" error.

What's the debug message detail on that? (not-negotiated?)

> Please see the attached code and the function call log.
> The magic happens in pcm_play_dma_init(), found_source() and feed_data().
> 
> Any idea what could be wrong? The timestamps on the GstBuffer?

That depends on the exact error, but if it's not-negotiated it's likely
got something to do with caps.

> Can I push arbitrary data size into the appsrc or must it be
> in sample size chunks?

That depends on what you're pushing into appsrc. If it's raw PCM audio,
it should be in chunks of bytes_per_sample * channels.


> The code is quite hacky at the moment as I want to get
> it up and running and then beautify it. Sorry:)

  audio_caps = gst_caps_new_simple("audio/x-raw-int",
      "width", G_TYPE_INT, (gint)16,
      "depth", G_TYPE_INT, (gint)16,
      "channels" ,G_TYPE_INT, (gint)2,
 -->  "signed",G_TYPE_INT,1, <--
      "rate", G_TYPE_INT, 44100,
      "endianness", G_TYPE_INT, (gint)1234,
      NULL);

The signed field should be of G_TYPE_BOOLEAN.

Cheers
 -Tim

[gst-embedded] Playback of raw audio on N900: Internal data flow error

[Thomas J. <to...@si...>]

Hello,

I'm currently trying to port rockbox to the Nokia N900. The rockbox 
application backend is using SDL for audio and I want to migrate
it to gstreamer.

I tried all kind of ways to push the raw audio via an "appsrc" to "playbin2".
The result is always an "Internal data flow" error.

Please see the attached code and the function call log.
The magic happens in pcm_play_dma_init(), found_source() and feed_data().

Any idea what could be wrong? The timestamps on the GstBuffer?

Can I push arbitrary data size into the appsrc or must it be
in sample size chunks?

The code is quite hacky at the moment as I want to get
it up and running and then beautify it. Sorry:)

Thanks in advance,
Thomas

Re: [gst-embedded] Unable to stream MPEG4 or H.264 from BeagleBoard

[Andrey N. <and...@ya...>]

Hi Matthew,

>gst-launch -v v4l2src ! TIVidenc1 codecName=h264enc
>engineName=codecServer ! rtph264pay pt=96 ! udpsink host=<DEST> port=5000

I think there is missing colorconversion element (ffmpegcolorspace) before 
TIVidenc1. Also, it is necessary to specify colorspace (fourcc) explicitly. You 
can take a look at this thread which discuss similar issue: 

https://gstreamer.ti.com/gf/project/gstreamer_ti/forum/?_forum_action=ForumMessageBrowse&thread_id=3251&action=ForumBrowse&forum_id=187


In addition you can take a look at some example pipelines I am using in my 
project:
http://www.gitorious.org/veter/veter/blobs/master/misc/car.config#line36

HTH,
Andrey.

----- Original Message ----
From: Matthew Braun <mj...@le...>
To: gst...@li...
Sent: Thu, December 9, 2010 5:29:15 PM
Subject: [gst-embedded] Unable to stream MPEG4 or H.264 from BeagleBoard

Greetings, and apologies in advance if my question is excessively simple,
although my searches have so far proven futile.

I'm running Angstrom on a BeagleBoard C (OMAP3530) with the gstreamer-ti
package (GStreamer 0.10.30) installed. At boot the following commands are
run:

cd /usr/share/ti/gst/omap3530/
./loadmodules.sh
export GST_REGISTRY=/tmp/gst_registry.bin
export LD_LIBRARY_PATH=/usr/lib
export GST_PLUGIN_PATH=/usr/lib/gstreamer-0.10
export PATH=/usr/bin:$PATH
cat /dev/zero > /dev/fb2 2> /dev/null


Running the command

gst-launch v4l2src ! video/x-raw-yuv,width=640,height=480 !
ffmpegcolorspace ! jpegenc ! multipartmux ! udpsink host=<DEST> port=5000

Succeeds (I can open VLC on <DEST> and view the stream, although it's
pretty choppy). However, I'd like to do an MPEG4 stream instead. Following
the example from
http://processors.wiki.ti.com/index.php/Example_GStreamer_Pipelines#OMAP35x
, I tried:

gst-launch-0.10  v4l2src ! 'video/x-raw-yuv,width=640,height=480' !
ffenc_mpeg4 ! rtpmp4vpay ! udpsink host=169.254.12.28 port=5000 -v

But I get

Setting pipeline to PAUSED ...
/GstPipeline:pipeline0/GstV4l2Src:v4l2src0.GstPad:src: caps =
video/x-raw-yuv, format=(fourcc)I420, width=(int)640, height=(int)480,
framerate=(fraction)30/1
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps =
video/x-raw-yuv, format=(fourcc)I420, width=(int)640, height=(int)480,
framerate=(fraction)30/1
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:sink: caps =
video/x-raw-yuv, format=(fourcc)I420, width=(int)640, height=(int)480,
framerate=(fraction)30/1
/GstPipeline:pipeline0/ffenc_mpeg4:ffenc_mpeg40.GstPad:src: caps =
video/mpeg, width=(int)640, height=(int)480, framerate=(fraction)30/1,
mpegversion=(int)4, systemstream=(boolean)false
/GstPipeline:pipeline0/ffenc_mpeg4:ffenc_mpeg40.GstPad:sink: caps =
video/x-raw-yuv, format=(fourcc)I420, width=(int)640, height=(int)480,
framerate=(fraction)30/1
/GstPipeline:pipeline0/GstRtpMP4VPay:rtpmp4vpay0.GstPad:sink: caps =
video/mpeg, width=(int)640, height=(int)480, framerate=(fraction)30/1,
mpegversion=(int)4, systemstream=(boolean)false
/GstPipeline:pipeline0/GstRtpMP4VPay:rtpmp4vpay0.GstPad:src: caps =
application/x-rtp, media=(string)video, clock-rate=(int)90000,
encoding-name=(string)MP4V-ES, profile-level-id=(string)1,
config=(string)000001b001000001b58913000001000000012000c48d8800f514043c1463
000001b24c61766335322e39372e32, payload=(int)96, ssrc=(uint)1464948709,
clock-base=(uint)3097574894, seqnum-base=(uint)47019
/GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps =
application/x-rtp, media=(string)video, clock-rate=(int)90000,
encoding-name=(string)MP4V-ES, profile-level-id=(string)1,
config=(string)000001b001000001b58913000001000000012000c48d8800f514043c1463
000001b24c61766335322e39372e32, payload=(int)96, ssrc=(uint)1464948709,
clock-base=(uint)3097574894, seqnum-base=(uint)47019
libv4l2: error converting / decoding frame data: v4l-convert: error
parsing JPEG header: Not a JPG file ?
Caught SIGSEGV accessing address 0x4
unable to fork gdb: Cannot allocate memory
Spinning.  Please run 'gdb gst-launch 1582' to continue debugging, Ctrl-C
to quit, or Ctrl-\ to dump core.



I assume this is related to the error parsing the JPEG header? The v4l2
source is a Logitech C310 USB webcam.

Trying to encode as h.264 instead, from the same examples, I use:
gst-launch -v v4l2src ! TIVidenc1 codecName=h264enc
engineName=codecServer ! rtph264pay pt=96 ! udpsink host=<DEST> port=5000
-v
But I get:
Setting pipeline to PAUSED ...
ERROR: Pipeline doesn't want to pause.
ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: Could not
negotiate format
Additional debug info:
  gstbasesrc.c(2755): gst_base_src_start ():
/GstPipeline:pipeline0/GstV4l2Src:v4l2src0:
Check your filtered caps, if any
Setting pipeline to NULL ...
Freeing pipeline ...


Cat /proc/cmdline gives:
console=ttyS2,115200n8 console=tty0 root=/dev/mmcblk0p2 rw rootwait
rootdelay=2 mem=80M video=omapfb:vram:2M,vram:4M



Any suggestions as to what I might be missing or something I should be
trying instead to get MPEG4 or H.264 streaming to work?

Thank you very much for any guidance or suggestions!


--
Matthew Braun
mj...@le...





------------------------------------------------------------------------------
_______________________________________________
Gstreamer-embedded mailing list
Gst...@li...
https://lists.sourceforge.net/lists/listinfo/gstreamer-embedded


      

[gst-embedded] Error parsing JPEG header: Not a JPG file

[Matthew B. <mj...@le...>]

Greetings, and apologies in advance if my question is excessively simple,
although my searches have so far proven futile.

I'm running Angstrom on a BeagleBoard C (OMAP3530) with the gstreamer-ti
package (GStreamer 0.10.30) installed. At boot the following commands are
run:

 cd /usr/share/ti/gst/omap3530/
 ./loadmodules.sh
 export GST_REGISTRY=/tmp/gst_registry.bin
 export LD_LIBRARY_PATH=/usr/lib
 export GST_PLUGIN_PATH=/usr/lib/gstreamer-0.10
 export PATH=/usr/bin:$PATH
 cat /dev/zero > /dev/fb2 2> /dev/null


Running the command

 gst-launch v4l2src ! video/x-raw-yuv,width=640,height=480 !
ffmpegcolorspace ! jpegenc ! multipartmux ! udpsink host=<DEST> port=5000

Succeeds (I can open VLC on <DEST> and view the stream, although it's
pretty choppy). However, I'd like to do an MPEG4 stream instead. Following
the example from 
http://processors.wiki.ti.com/index.php/Example_GStreamer_Pipelines#OMAP35x
 , I tried:

 gst-launch-0.10  v4l2src ! 'video/x-raw-yuv,width=640,height=480' !
ffenc_mpeg4 ! rtpmp4vpay ! udpsink host=169.254.12.28 port=5000 -v

But I get 

 Setting pipeline to PAUSED ...
 /GstPipeline:pipeline0/GstV4l2Src:v4l2src0.GstPad:src: caps =
video/x-raw-yuv, format=(fourcc)I420, width=(int)640, height=(int)480,
framerate=(fraction)30/1
 Pipeline is live and does not need PREROLL ...
 Setting pipeline to PLAYING ...
 New clock: GstSystemClock
 /GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps =
video/x-raw-yuv, format=(fourcc)I420, width=(int)640, height=(int)480,
framerate=(fraction)30/1
 /GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:sink: caps =
video/x-raw-yuv, format=(fourcc)I420, width=(int)640, height=(int)480,
framerate=(fraction)30/1
 /GstPipeline:pipeline0/ffenc_mpeg4:ffenc_mpeg40.GstPad:src: caps =
video/mpeg, width=(int)640, height=(int)480, framerate=(fraction)30/1,
mpegversion=(int)4, systemstream=(boolean)false
 /GstPipeline:pipeline0/ffenc_mpeg4:ffenc_mpeg40.GstPad:sink: caps =
video/x-raw-yuv, format=(fourcc)I420, width=(int)640, height=(int)480,
framerate=(fraction)30/1
 /GstPipeline:pipeline0/GstRtpMP4VPay:rtpmp4vpay0.GstPad:sink: caps =
video/mpeg, width=(int)640, height=(int)480, framerate=(fraction)30/1,
mpegversion=(int)4, systemstream=(boolean)false
 /GstPipeline:pipeline0/GstRtpMP4VPay:rtpmp4vpay0.GstPad:src: caps =
application/x-rtp, media=(string)video, clock-rate=(int)90000,
encoding-name=(string)MP4V-ES, profile-level-id=(string)1,
config=(string)000001b001000001b58913000001000000012000c48d8800f514043c1463
000001b24c61766335322e39372e32, payload=(int)96, ssrc=(uint)1464948709,
clock-base=(uint)3097574894, seqnum-base=(uint)47019
 /GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps =
application/x-rtp, media=(string)video, clock-rate=(int)90000,
encoding-name=(string)MP4V-ES, profile-level-id=(string)1,
config=(string)000001b001000001b58913000001000000012000c48d8800f514043c1463
000001b24c61766335322e39372e32, payload=(int)96, ssrc=(uint)1464948709,
clock-base=(uint)3097574894, seqnum-base=(uint)47019
 libv4l2: error converting / decoding frame data: v4l-convert: error
parsing JPEG header: Not a JPG file ?
 Caught SIGSEGV accessing address 0x4
 unable to fork gdb: Cannot allocate memory
 Spinning.  Please run 'gdb gst-launch 1582' to continue debugging, Ctrl-C
to quit, or Ctrl-\ to dump core.



I assume this is related to the error parsing the JPEG header? The v4l2
source is a Logitech C310 USB webcam.

Trying to encode as h.264 instead, from the same examples, I use:
 
 gst-launch -v v4l2src ! TIVidenc1 codecName=h264enc
engineName=codecServer ! rtph264pay pt=96 ! udpsink host=<DEST> port=5000
-v
 
But I get:
 
 Setting pipeline to PAUSED ...
 ERROR: Pipeline doesn't want to pause.
 ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: Could not
negotiate format
 Additional debug info:
  gstbasesrc.c(2755): gst_base_src_start ():
/GstPipeline:pipeline0/GstV4l2Src:v4l2src0:
 Check your filtered caps, if any
 Setting pipeline to NULL ...
 Freeing pipeline ...


 Cat /proc/cmdline gives:
 
 console=ttyS2,115200n8 console=tty0 root=/dev/mmcblk0p2 rw rootwait
rootdelay=2 mem=80M video=omapfb:vram:2M,vram:4M



Any suggestions as to what I might be missing or something I should be
trying instead to get MPEG4 or H.264 streaming to work?

Thank you very much for any guidance or suggestions!

--
Matthew Braun
mj...@le...

[gst-embedded] Unable to stream MPEG4 or H.264 from BeagleBoard

[Matthew B. <mj...@le...>]

Greetings, and apologies in advance if my question is excessively simple,
although my searches have so far proven futile.

I'm running Angstrom on a BeagleBoard C (OMAP3530) with the gstreamer-ti
package (GStreamer 0.10.30) installed. At boot the following commands are
run:

cd /usr/share/ti/gst/omap3530/
./loadmodules.sh
export GST_REGISTRY=/tmp/gst_registry.bin
export LD_LIBRARY_PATH=/usr/lib
export GST_PLUGIN_PATH=/usr/lib/gstreamer-0.10
export PATH=/usr/bin:$PATH
cat /dev/zero > /dev/fb2 2> /dev/null


Running the command

gst-launch v4l2src ! video/x-raw-yuv,width=640,height=480 !
ffmpegcolorspace ! jpegenc ! multipartmux ! udpsink host=<DEST> port=5000

Succeeds (I can open VLC on <DEST> and view the stream, although it's
pretty choppy). However, I'd like to do an MPEG4 stream instead. Following
the example from
http://processors.wiki.ti.com/index.php/Example_GStreamer_Pipelines#OMAP35x
, I tried:

gst-launch-0.10  v4l2src ! 'video/x-raw-yuv,width=640,height=480' !
ffenc_mpeg4 ! rtpmp4vpay ! udpsink host=169.254.12.28 port=5000 -v

But I get

Setting pipeline to PAUSED ...
/GstPipeline:pipeline0/GstV4l2Src:v4l2src0.GstPad:src: caps =
video/x-raw-yuv, format=(fourcc)I420, width=(int)640, height=(int)480,
framerate=(fraction)30/1
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps =
video/x-raw-yuv, format=(fourcc)I420, width=(int)640, height=(int)480,
framerate=(fraction)30/1
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:sink: caps =
video/x-raw-yuv, format=(fourcc)I420, width=(int)640, height=(int)480,
framerate=(fraction)30/1
/GstPipeline:pipeline0/ffenc_mpeg4:ffenc_mpeg40.GstPad:src: caps =
video/mpeg, width=(int)640, height=(int)480, framerate=(fraction)30/1,
mpegversion=(int)4, systemstream=(boolean)false
/GstPipeline:pipeline0/ffenc_mpeg4:ffenc_mpeg40.GstPad:sink: caps =
video/x-raw-yuv, format=(fourcc)I420, width=(int)640, height=(int)480,
framerate=(fraction)30/1
/GstPipeline:pipeline0/GstRtpMP4VPay:rtpmp4vpay0.GstPad:sink: caps =
video/mpeg, width=(int)640, height=(int)480, framerate=(fraction)30/1,
mpegversion=(int)4, systemstream=(boolean)false
/GstPipeline:pipeline0/GstRtpMP4VPay:rtpmp4vpay0.GstPad:src: caps =
application/x-rtp, media=(string)video, clock-rate=(int)90000,
encoding-name=(string)MP4V-ES, profile-level-id=(string)1,
config=(string)000001b001000001b58913000001000000012000c48d8800f514043c1463
000001b24c61766335322e39372e32, payload=(int)96, ssrc=(uint)1464948709,
clock-base=(uint)3097574894, seqnum-base=(uint)47019
/GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps =
application/x-rtp, media=(string)video, clock-rate=(int)90000,
encoding-name=(string)MP4V-ES, profile-level-id=(string)1,
config=(string)000001b001000001b58913000001000000012000c48d8800f514043c1463
000001b24c61766335322e39372e32, payload=(int)96, ssrc=(uint)1464948709,
clock-base=(uint)3097574894, seqnum-base=(uint)47019
libv4l2: error converting / decoding frame data: v4l-convert: error
parsing JPEG header: Not a JPG file ?
Caught SIGSEGV accessing address 0x4
unable to fork gdb: Cannot allocate memory
Spinning.  Please run 'gdb gst-launch 1582' to continue debugging, Ctrl-C
to quit, or Ctrl-\ to dump core.



I assume this is related to the error parsing the JPEG header? The v4l2
source is a Logitech C310 USB webcam.

Trying to encode as h.264 instead, from the same examples, I use:
gst-launch -v v4l2src ! TIVidenc1 codecName=h264enc
engineName=codecServer ! rtph264pay pt=96 ! udpsink host=<DEST> port=5000
-v
But I get:
Setting pipeline to PAUSED ...
ERROR: Pipeline doesn't want to pause.
ERROR: from element /GstPipeline:pipeline0/GstV4l2Src:v4l2src0: Could not
negotiate format
Additional debug info:
  gstbasesrc.c(2755): gst_base_src_start ():
/GstPipeline:pipeline0/GstV4l2Src:v4l2src0:
Check your filtered caps, if any
Setting pipeline to NULL ...
Freeing pipeline ...


Cat /proc/cmdline gives:
console=ttyS2,115200n8 console=tty0 root=/dev/mmcblk0p2 rw rootwait
rootdelay=2 mem=80M video=omapfb:vram:2M,vram:4M



Any suggestions as to what I might be missing or something I should be
trying instead to get MPEG4 or H.264 streaming to work?

Thank you very much for any guidance or suggestions!


--
Matthew Braun
mj...@le...