[gst-devel] gstreamer-devel Digest, Vol 36, Issue 94
Suresh Choudary
sikkim.suresh at gmail.com
Sun May 31 16:16:26 CEST 2009
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 at 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 at 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 at lists.sourceforge.net>
> Message-ID: <200905301657.26757.lists at 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 at 163.com>
> Subject: Re: [gst-devel] Problem of transporting the ts stream over
> To: "Zhiqiang Liu" <liuzq2002 at 126.com>
> Cc: gstreamer-devel <gstreamer-devel at lists.sourceforge.net>
> Message-ID:
> <59890.64791243734463071.JavaMail.coremail at bj163app15.163.com>
>
> WLAN802.11
> MIME-Version: 1.0
> Content-Type: multipart/alternative;
> boundary="----=_Part_17596_27235653.1243734463069"
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> X-Priority: 3
> X-Mailer: Coremail Webmail Server Version XT2_snapshot build
> 090513(7592.2351.2332) Copyright (c) 2002-2009 www.mailtech.cn 163com
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> 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" <liuzq2002 at 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" <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 <bisht.sudarshan at 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 at lists.sourceforge.net>
> Message-ID:
> <785339900905302228n1000adf8pcfd2aec674bf03cb at 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 at 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.
> >
> >
> >
> > ------------------------------
> > Invite your mail contacts to join your friends list with Windows Live
> > Spaces. It's easy! Try it!<
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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, &
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> > _______________________________________________
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> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
> >
> >
>
>
> --
> Regards,
>
> Sudarshan Bisht
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> ------------------------------
>
> Message: 4
> Date: Sun, 31 May 2009 11:18:51 +0530
> From: sudarshan bisht <bisht.sudarshan at gmail.com>
> Subject: Re: [gst-devel] PLAy->PAUSE Issue with alsasink
> To: Discussion of the development of GStreamer
> <gstreamer-devel at lists.sourceforge.net>
> Message-ID:
> <785339900905302248x33561748ve5dc658be3c7ac00 at 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 at 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
> > _______________________________________________
> > gstreamer-devel mailing list
> > gstreamer-devel at lists.sourceforge.net
> > 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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>
> End of gstreamer-devel Digest, Vol 36, Issue 94
> ***********************************************
>
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