[gst-devel] gstreamer-devel Digest, Vol 36, Issue 94
sudarshan bisht
bisht.sudarshan at gmail.com
Mon Jun 1 06:47:02 CEST 2009
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 <sikkim.suresh at gmail.com>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, <
> 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
>> -------------- next part --------------
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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"
>> 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" <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
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>> > Meet
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>> > _______________________________________________
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>> > gstreamer-devel at lists.sourceforge.net
>> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>> >
>> >
>>
>>
>> --
>> Regards,
>>
>> Sudarshan Bisht
>> -------------- next part --------------
>> An HTML attachment was scrubbed...
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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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>> ------------------------------
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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
>>
>> ------------------------------
>>
>> _______________________________________________
>> gstreamer-devel mailing list
>> gstreamer-devel at lists.sourceforge.net
>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>>
>>
>> End of gstreamer-devel Digest, Vol 36, Issue 94
>> ***********************************************
>>
>
>
>
> ------------------------------------------------------------------------------
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> is a gathering of tech-side developers & brand creativity professionals.
> Meet
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> 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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--
Regards,
Sudarshan Bisht
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