Hi Sudarshan,
 
Thanks for the response. Navtest it just a simple plugin that allows console input such as p for PAUSE, R for rewind by some configured number of seconds,  F for forward and so on.
 
It just passes the key events as PIPELINE state commands. The chain function in the navtest is dummy and just passes the incoming buffers to next element as it is.
 
It is being used by us just for the sake of simplicity and ease of debugging various scenarious in various combinations of plugins and fileformats.
 
BR,
Suresh


 
On Sun, May 31, 2009 at 11:18 AM, <gstreamer-devel-request@lists.sourceforge.net> wrote:
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Today's Topics:

  1. Re: Dinamically add clients to multiudpsink: why and      how use
     a signal??? (MailingList SVR)
  2. Re: Problem of transporting the ts stream over (Volter Yen)
  3. Re: How to save a stream from a network into a file
     (sudarshan bisht)
  4. Re: PLAy->PAUSE Issue with alsasink (sudarshan bisht)


----------------------------------------------------------------------

Message: 1
Date: Sat, 30 May 2009 16:57:26 +0200
From: MailingList SVR <lists@svrinformatica.it>
Subject: Re: [gst-devel] Dinamically add clients to multiudpsink: why
       and     how use a signal???
To: Discussion of the development of GStreamer
       <gstreamer-devel@lists.sourceforge.net>
Message-ID: <200905301657.26757.lists@svrinformatica.it>
Content-Type: text/plain; charset="iso-8859-15"

In data sabato 30 maggio 2009 15:49:32, MailingList SVR ha scritto:
: > Hi all,
>
> there is something not much clear to me about multiupdsink: I would like to dinamycally add clients to multiudpsink, based on the documentation (http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-multiudpsink.html) there are:
>
> 1) a clients property I can populate with the desidered clients, ok is fine
> 2) an "add" signal???? But how add clients using a signal?
>
> I tried to modify the clients property while the pipeline is running but this didn't work, so the only way if one is to use the add signal but I don't know how to use a signal to add a client can you give me some examples please? I'm using the python bindings,
>
> thanks
> Nicola
>

Ok solved,

thanks
Nicola
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Message: 2
Date: Sun, 31 May 2009 09:47:43 +0800 (CST)
From: "Volter Yen" <volter619@163.com>
Subject: Re: [gst-devel] Problem of transporting the ts stream over
To: "Zhiqiang Liu" <liuzq2002@126.com>
Cc: gstreamer-devel <gstreamer-devel@lists.sourceforge.net>
Message-ID:
       <59890.64791243734463071.JavaMail.coremail@bj163app15.163.com>

 WLAN802.11
MIME-Version: 1.0
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=BF=EC=BD=DD=BB=D8=B8=B4=B8=F8=A3=BA"=B9=E3=D6=DD=CA=FD=BE=DD=D6=D0=D0=C4"=
=20

=D4=DA2009-05-28=A3=AC"Zhiqiang Liu" <liuzq2002@126.com> =D0=B4=B5=C0=A3=BA


Hi ScreenName01,
=20
Thanks for your help:-)=20
=20
The "ideal environment" refer to transport the udp packets in the wire comm=
unication. In this case, The possiblity of losing the packets is very small=
.

There seems to be no encryption problem since we can send the raw mpeg stre=
ams over the air to the target and play on it.=20

It's really an unusual problem since we know that the the underlying medium=
 is hidden to the protocol. The only possibly problem can occur in the MAC =
layer. The WLAN may lose some packets (About 10% packets are lost). But in =
the wire communication almost very packets are delivered normally. The prob=
lem may be related to the ts stream format. That's because it may be hard t=
o play an ts stream when some packets are lost.

Thanks for your suggestion. I will try to analyse the traffic using wiresha=
rk.

I would like to keep in touch with you. When we get any progress, I will co=
ntact you.

=20

Best regards,

Zhiqiang Liu

ScreenName01 wrote:
>Hi Zhiqiang,
>
>  I'm unclear of what the problem is.  What is an "ideal environment" for
>instance?
>
>  The underlying medium -- be it ethernet or wifi -- is transparent.  The
>medium is hidden to the protocol and is handled by the OS in most cases

------=_Part_17596_27235653.1243734463069
Content-Type: text/html; charset=gbk
Content-Transfer-Encoding: quoted-printable

=BF=EC=BD=DD=BB=D8=B8=B4=B8=F8=A3=BA"=B9=E3=D6=DD=CA=FD=BE=DD=D6=D0=D0=C4" =
<admin5 br=3D""><br><br>=D4=DA2009-05-28=A3=AC"Zhiqiang Liu" &lt;liuzq2002@=
126.com&gt; =D0=B4=B5=C0=A3=BA<br> <BLOCKQUOTE id=3D"isReplyContent" style=
=3D"PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px sol=
id"><div><br>Hi ScreenName01,</div>
<div>&nbsp;</div>
<div>Thanks for your help:-) </div>
<div>&nbsp;</div>
<div>The "ideal environment" refer to transport the udp packets in the wire=
 communication. In this case, The possiblity&nbsp;of losing&nbsp;the packet=
s is very small.</div>
<div></div>
<p>There seems to be no encryption problem since we can send the raw mpeg s=
treams over the air to the target and play on it.&nbsp;</p>
<p>It's really an unusual problem since we know that the the underlying med=
ium is hidden to the protocol. The only possibly problem&nbsp;can&nbsp;occu=
r in the MAC layer. The WLAN may lose some packets (About 10% packets are l=
ost).&nbsp;But in the wire communication almost very packets are delivered =
normally. The problem may be related to the ts stream format. That's becaus=
e it may be hard to play an ts stream when&nbsp;some packets are lost.</p>
<p>Thanks for your suggestion. I will try to analyse the&nbsp;traffic&nbsp;=
using&nbsp;wireshark.</p>
<p>I would like to keep in touch with you.&nbsp;When we get&nbsp;any progre=
ss, I will contact you.</p>
<p>&nbsp;</p>
<p>Best regards,</p>
<p>Zhiqiang Liu</p><pre>ScreenName01 wrote:
&gt;Hi Zhiqiang,
&gt;
&gt;  I'm unclear of what the problem is.  What is an "ideal environment" f=
or
&gt;instance?
&gt;
&gt;  The underlying medium -- be it ethernet or wifi -- is transparent.  T=
he
&gt;medium is hidden to the protocol and is handled by the OS in most cases
</pre></BLOCKQUOTE></admin5><br><!-- footer --><br><span title=3D"neteasefo=
oter"/><hr/>
<a href=3D"http://512.mail.163.com/mailstamp/stamp/dz/activity.do?from=3Dfo=
oter">=B4=A9=D4=BD=B5=D8=D5=F0=B4=F8 =BC=CD=C4=EE=E3=EB=B4=A8=B5=D8=D5=F0=
=D2=BB=D6=DC=C4=EA</a>
</span>
------=_Part_17596_27235653.1243734463069--




------------------------------

Message: 3
Date: Sun, 31 May 2009 10:58:31 +0530
From: sudarshan bisht <bisht.sudarshan@gmail.com>
Subject: Re: [gst-devel] How to save a stream from a network into a
       file
To: Discussion of the development of GStreamer
       <gstreamer-devel@lists.sourceforge.net>
Message-ID:
       <785339900905302228n1000adf8pcfd2aec674bf03cb@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi,          Hi ,,
        Try providing caps between  rtph263pdepay and avimux .



On Sat, May 30, 2009 at 8:28 PM, Zelalem Sintayehu <zelalems@hotmail.com>wrote:

>  Hi, I was trying to transfer video and audio using network. I used teh
> examples from the net to do that and succeeded. But now I wanted to save the
> stream into file and faced with some problem. Please look at the following
> command:
>
> gst-launch-0.10 udpsrc port=5000 caps="application/x-rtp,
> media=(string)video,clock-rate=(int)90000, encoding-name=(string)H263-1998"
> num-buffers=5000 ! queue ! rtph263pdepay ! ffdec_h263 ! xvimagesink   -----
> this is what i used to accept and display a video stream.
>
> So, to save the stream into a file I changed the last two elements (the
> ffmpeg decoder and xvimake sink). I thought that since the packet coming
> from the other machine is already encoded in h263p codec, replacing these
> two elements  with the following elements would solve my problem: I used
> these elments: avimux ! filesink location=testnet.avi . That is, i connected
> the rtph263pdepay element to the avimux element and to the file sink element
> sequentially as follows.
>
>  gst-launch-0.10 udpsrc port=5000 caps="application/x-rtp,
> media=(string)video,clock-rate=(int)90000, encoding-name=(string)H263-1998"
> num-buffers=5000 ! queue ! rtph263pdepay ! avimux ! filesink
> location=test.avi
>
> But I got an error, that says: streaming task paused, reason not-negotiated
> (-4)
>
> Please help me on how I can save a stream.
>
> Thank you.
>
> - Zelalem S.
>
>
>
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>


--
Regards,

Sudarshan Bisht
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Message: 4
Date: Sun, 31 May 2009 11:18:51 +0530
From: sudarshan bisht <bisht.sudarshan@gmail.com>
Subject: Re: [gst-devel] PLAy->PAUSE Issue with alsasink
To: Discussion of the development of GStreamer
       <gstreamer-devel@lists.sourceforge.net>
Message-ID:
       <785339900905302248x33561748ve5dc658be3c7ac00@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi ,       I have few questions .

      Why are you using navtest plugin to perform PLAY/PAUSE/SEEK ? because
that can be done using your application also.

  And what is the implementation of navtest i mean what exactly you are
doing in that plugin  ?




On Sat, May 30, 2009 at 6:57 PM, Suresh Choudary <sikkim.suresh@gmail.com>wrote:

> Dear All,
>
> I am using the following pipeline with gstreamer version 0.10.22 and latest
> plugins.
>
> gst-launch filesrc location=/home/testh263.3gp ! qtdemux name=demux
> demux.audio_00 ! queue ! amrdecoder ! navtest ! alsasink demux.video ! queue
> ! h263decoder ! v4l2sink
>
> where navtest is a simple plugin which allows user to PLAY/PAUSE/SEEK.
>
> Overall the pipeline is as follows from application point of view.
>
>                                  |----------> queue ---> amrdecoder
> --->alsasink
> filesrc--->qtdemux   ----|
>
> |----------->queue---->h263decoder--->v4l2sink
>
> Where I am using the open source alsasink and custom decoders. When I try
> to set the pipeline to PAUSED state, some times (1 out of 10 times) all the
> components can transition to PAUSED state, but alsasink sends a ASYNC
> notification, but never commits to paused state. (As the part log below
> shows the same.I have enabled only basesink logs)
>
>
>
> --------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
>
> 0:02:07.538391114   865    0xcfdd0 DEBUG             basesink
> /home/root/x2//middleware/multimedia-framework/media-service/gstreamer/gstreamer-0.10.22/./libs/gst/base/gstbasesink.c:2911:gst_base_sink_chain_unlocked:<avsysvideosink0>
> got times start: 0:00:23.648648648, end: 0:00:23.690357023
>
> 0:02:07.538726807   865    0xcfdd0 DEBUG             basesink
> /home/root/x2//middleware/multimedia-framework/media-service/gstreamer/gstreamer-0.10.22/./libs/gst/base/gstbasesink.c:1534:gst_base_sink_get_sync_times:<avsysvideosink0>
> got times start: 0:00:23.648648648, stop: 0:00:23.690357023, do_sync 1
>
> 0:02:07.538970948   865    0xcfdd0 DEBUG             basesink
> /home/root/x2//middleware/multimedia-framework/media-service/gstreamer/gstreamer-0.10.22/./libs/gst/base/gstbasesink.c:1984:gst_base_sink_do_sync:<avsysvideosink0>
> possibly waiting for clock to reach 0:00:23.648648648, adjusted
> 0:00:23.648648648
>
> 0:02:07.590026856   865    0xcfe80 DEBUG             basesink
> /home/root/x2//middleware/multimedia-framework/media-service/gstreamer/gstreamer-0.10.22/./libs/gst/base/gstbasesink.c:4099:gst_base_sink_change_state:<avsysvideosink0>
> PLAYING to PAUSED
>
> 0:02:07.611236573   865    0xcfe80 DEBUG             basesink
> /home/root/x2//middleware/multimedia-framework/media-service/gstreamer/gstreamer-0.10.22/./libs/gst/base/gstbasesink.c:2846:gst_base_sink_needs_preroll:<avsysvideosink0>
> have_preroll: 0, EOS: 0 => needs preroll: 1
>
> 0:02:07.611511231   865    0xcfe80 DEBUG             basesink
> /home/root/x2//middleware/multimedia-framework/media-service/gstreamer/gstreamer-0.10.22/./libs/gst/base/gstbasesink.c:4131:gst_base_sink_change_state:<avsysvideosink0>
> PLAYING to PAUSED, we are not prerolled
>
> 0:02:07.611694336   865    0xcfe80 DEBUG             basesink
> /home/root/x2//middleware/multimedia-framework/media-service/gstreamer/gstreamer-0.10.22/./libs/gst/base/gstbasesink.c:4135:gst_base_sink_change_state:<avsysvideosink0>
> doing async state change
>
> 0:02:07.612030030   865    0xcfe80 DEBUG             basesink
> /home/root/x2//middleware/multimedia-framework/media-service/gstreamer/gstreamer-0.10.22/./libs/gst/base/gstbasesink.c:4144:gst_base_sink_change_state:<avsysvideosink0>
> rendered: 13, dropped: 53
>
> [gst_avsysvideosink_change_state:835]GST_STATE_CHANGE_PLAYING_TO_PAUSED
>
> 0:02:07.612487793   865    0xcfdd0 DEBUG             basesink
> /home/root/x2//middleware/multimedia-framework/media-service/gstreamer/gstreamer-0.10.22/./libs/gst/base/gstbasesink.c:1990:gst_base_sink_do_sync:<avsysvideosink0>
> clock returned 2
>
> 0:02:07.612731934   865    0xcfdd0 DEBUG             basesink
> /home/root/x2//middleware/multimedia-framework/media-service/gstreamer/gstreamer-0.10.22/./libs/gst/base/gstbasesink.c:2003:gst_base_sink_do_sync:<avsysvideosink0>
> unscheduled, waiting some more
>
> 0:02:07.612915039   865    0xcfdd0 DEBUG             basesink
> /home/root/x2//middleware/multimedia-framework/media-service/gstreamer/gstreamer-0.10.22/./libs/gst/base/gstbasesink.c:1943:gst_base_sink_do_sync:<avsysvideosink0>
> prerolling object 0xe2ad8
>
> 0:02:07.613098145   865    0xcfdd0 DEBUG             basesink
> /home/root/x2//middleware/multimedia-framework/media-service/gstreamer/gstreamer-0.10.22/./libs/gst/base/gstbasesink.c:1357:gst_base_sink_commit_state:<avsysvideosink0>
> commiting state to PAUSED
>
> 0:02:07.613281250   865    0xcfdd0 DEBUG             basesink
> /home/root/x2//middleware/multimedia-framework/media-service/gstreamer/gstreamer-0.10.22/./libs/gst/base/gstbasesink.c:1382:gst_base_sink_commit_state:<avsysvideosink0>
> posting PAUSED state change message
>
> 0:02:07.614196778   865    0xcfdd0 DEBUG             basesink
> /home/root/x2//middleware/multimedia-framework/media-service/gstreamer/gstreamer-0.10.22/./libs/gst/base/gstbasesink.c:1388:gst_base_sink_commit_state:<avsysvideosink0>
> posting async-done message
>
> 0:02:07.614532471   865    0xcfdd0 DEBUG             basesink
> /home/root/x2//middleware/multimedia-framework/media-service/gstreamer/gstreamer-0.10.22/./libs/gst/base/gstbasesink.c:1732:gst_base_sink_wait_preroll:<avsysvideosink0>
> waiting in preroll for flush or PLAYING
>
> *0:02:07.620910645   865    0xcfe80 DEBUG             basesink
> /home/root/x2//middleware/multimedia-framework/media-service/gstreamer/gstreamer-0.10.22/./libs/gst/base/gstbasesink.c:4099:gst_base_sink_change_state:<alsasink0>
> PLAYING to PAUSED*
>
> *0:02:07.621154785   865    0xcfe80 DEBUG             basesink
> /home/root/x2//middleware/multimedia-framework/media-service/gstreamer/gstreamer-0.10.22/./libs/gst/base/gstbasesink.c:2846:gst_base_sink_needs_preroll:<alsasink0>
> have_preroll: 0, EOS: 0 => needs preroll: 1*
>
> *0:02:07.653625489   865    0xcfe80 DEBUG             basesink
> /home/root/x2//middleware/multimedia-framework/media-service/gstreamer/gstreamer-0.10.22/./libs/gst/base/gstbasesink.c:4131:gst_base_sink_change_state:<alsasink0>
> PLAYING to PAUSED, we are not prerolled*
>
> *0:02:07.653900147   865    0xcfe80 DEBUG             basesink
> /home/root/x2//middleware/multimedia-framework/media-service/gstreamer/gstreamer-0.10.22/./libs/gst/base/gstbasesink.c:4135:gst_base_sink_change_state:<alsasink0>
> doing async state change*
>
> 0:02:07.654205323   865    0xcfe80 DEBUG             basesink
> /home/root/x2//middleware/multimedia-framework/media-service/gstreamer/gstreamer-0.10.22/./libs/gst/base/gstbasesink.c:4144:gst_base_sink_change_state:<alsasink0>
> rendered: 157, dropped: 0
>
> PAUSED
>
>
>
>
>
> But after this the audio sink (alsasink) can not commit the state to pause.
> I understand this happens because no more buffers are pushed by amrdecoder
> to alsasink but somehow the qtdemux is also blocked and sends no data to
> amrdecoder which may cause the sink to get one buffer and get prerolled and
> commit the state.
>
>
>
> I want to enquire if anyone of you have faced similar issue, and how to go
> about this issue. Please help me resolve this issue.
>
>
>
> BR,
>
> Suresh
>
>
>
>
> ------------------------------------------------------------------------------
> Register Now for Creativity and Technology (CaT), June 3rd, NYC. CaT
> is a gathering of tech-side developers & brand creativity professionals.
> Meet
> the minds behind Google Creative Lab, Visual Complexity, Processing, &
> iPhoneDevCamp as they present alongside digital heavyweights like Barbarian
> Group, R/GA, & Big Spaceship. http://p.sf.net/sfu/creativitycat-com
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> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>
>


--
Regards,

Sudarshan Bisht
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Register Now for Creativity and Technology (CaT), June 3rd, NYC. CaT
is a gathering of tech-side developers & brand creativity professionals. Meet
the minds behind Google Creative Lab, Visual Complexity, Processing, &
iPhoneDevCamp as they present alongside digital heavyweights like Barbarian
Group, R/GA, & Big Spaceship. http://p.sf.net/sfu/creativitycat-com

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