|
|
Hi Suresh ,
Could you please elaborate more on which Gstreamer API Navtest is using to send these events ? I think you have set properties with Navtest to receive the console input (P , R, F) .
On Sun, May 31, 2009 at 7:46 PM, Suresh Choudary <[hidden email]> wrote:
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, <[hidden email]> wrote:
Send gstreamer-devel mailing list submissions to
[hidden email]
To subscribe or unsubscribe via the World Wide Web, visit
https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
or, via email, send a message with subject or body 'help' to
[hidden email]
You can reach the person managing the list at
[hidden email]
When replying, please edit your Subject line so it is more specific
than "Re: Contents of gstreamer-devel digest..."
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 <[hidden email]>
Subject: Re: [gst-devel] Dinamically add clients to multiudpsink: why
and how use a signal???
To: Discussion of the development of GStreamer
<[hidden email]>
Message-ID: <[hidden email]>
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
-------------- next part --------------
An HTML attachment was scrubbed...
------------------------------
Message: 2
Date: Sun, 31 May 2009 09:47:43 +0800 (CST)
From: "Volter Yen" <[hidden email]>
Subject: Re: [gst-devel] Problem of transporting the ts stream over
To: "Zhiqiang Liu" <[hidden email]>
Cc: gstreamer-devel <[hidden email]>
Message-ID:
<[hidden email]>
WLAN802.11
MIME-Version: 1.0
Content-Type: multipart/alternative;
boundary="----=_Part_17596_27235653.1243734463069"
X-Originating-IP: [61.144.246.170]
X-Priority: 3
X-Mailer: Coremail Webmail Server Version XT2_snapshot build
090513(7592.2351.2332) Copyright (c) 2002-2009 www.mailtech.cn 163com
------=_Part_17596_27235653.1243734463069
Content-Type: text/plain; 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"=
=20
=D4=DA2009-05-28=A3=AC"Zhiqiang Liu" <[hidden email]> =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" <liuzq2002@=
126.com> =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> </div>
<div>Thanks for your help:-) </div>
<div> </div>
<div>The "ideal environment" refer to transport the udp packets in the wire=
communication. In this case, The possiblity of losing 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. </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 can occu=
r in the MAC layer. The WLAN may lose some packets (About 10% packets are l=
ost). 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 some packets are lost.</p>
<p>Thanks for your suggestion. I will try to analyse the traffic =
using wireshark.</p>
<p>I would like to keep in touch with you. When we get any progre=
ss, I will contact you.</p>
<p> </p>
<p>Best regards,</p>
<p>Zhiqiang Liu</p><pre>ScreenName01 wrote:
>Hi Zhiqiang,
>
> I'm unclear of what the problem is. What is an "ideal environment" f=
or
>instance?
>
> The underlying medium -- be it ethernet or wifi -- is transparent. T=
he
>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 <[hidden email]>
Subject: Re: [gst-devel] How to save a stream from a network into a
file
To: Discussion of the development of GStreamer
<[hidden email]>
Message-ID:
<[hidden email]>
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 <[hidden email]>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.
>
>
>
> ------------------------------
> Invite your mail contacts to join your friends list with Windows Live
> Spaces. It's easy! Try it!<http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx&mkt=en-us>
>
>
> ------------------------------------------------------------------------------
> 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
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>
>
--
Regards,
Sudarshan Bisht
-------------- next part --------------
An HTML attachment was scrubbed...
------------------------------
Message: 4
Date: Sun, 31 May 2009 11:18:51 +0530
From: sudarshan bisht <[hidden email]>
Subject: Re: [gst-devel] PLAy->PAUSE Issue with alsasink
To: Discussion of the development of GStreamer
<[hidden email]>
Message-ID:
<[hidden email]>
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 <[hidden email]>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
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>
>
--
Regards,
Sudarshan Bisht
-------------- next part --------------
An HTML attachment was scrubbed...
------------------------------
------------------------------------------------------------------------------
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
------------------------------
_______________________________________________
gstreamer-devel mailing list
[hidden email]
https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
End of gstreamer-devel Digest, Vol 36, Issue 94
***********************************************
------------------------------------------------------------------------------
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
_______________________________________________
gstreamer-devel mailing list
[hidden email]
https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
-- Regards, Sudarshan Bisht
------------------------------------------------------------------------------
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 _______________________________________________
gstreamer-devel mailing list
[hidden email]
https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
|
Locked
