<div>Hi Sudarshan,</div> <div> </div> <div>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.</div> <div> </div> <div>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.</div> <div> </div> <div>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.</div> <div> </div> <div>BR,</div> <div>Suresh</div> <div> </div> <div class="gmail_quote">On Sun, May 31, 2009 at 11:18 AM, <<a href="mailto:gstreamer-devel-request@lists.sourceforge.net">gstreamer-devel-request@lists.sourceforge.net</a>> wrote: <blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Send gstreamer-devel mailing list submissions to <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> To subscribe or unsubscribe via the World Wide Web, visit <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a> or, via email, send a message with subject or body 'help' to <a href="mailto:gstreamer-devel-request@lists.sourceforge.net">gstreamer-devel-request@lists.sourceforge.net</a> You can reach the person managing the list at <a href="mailto:gstreamer-devel-owner@lists.sourceforge.net">gstreamer-devel-owner@lists.sourceforge.net</a> 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 <<a href="mailto:lists@svrinformatica.it">lists@svrinformatica.it</a>> Subject: Re: [gst-devel] Dinamically add clients to multiudpsink: why and how use a signal??? To: Discussion of the development of GStreamer <<a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a>> Message-ID: <<a href="mailto:200905301657.26757.lists@svrinformatica.it">200905301657.26757.lists@svrinformatica.it</a>> 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 (<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-multiudpsink.html" target="_blank">http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-multiudpsink.html</a>) 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" <<a href="mailto:volter619@163.com">volter619@163.com</a>> Subject: Re: [gst-devel] Problem of transporting the ts stream over To: "Zhiqiang Liu" <<a href="mailto:liuzq2002@126.com">liuzq2002@126.com</a>> Cc: gstreamer-devel <<a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a>> Message-ID: <<a href="mailto:59890.64791243734463071.JavaMail.coremail@bj163app15.163.com">59890.64791243734463071.JavaMail.coremail@bj163app15.163.com</a>> 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 <a href="http://www.mailtech.cn/" target="_blank">www.mailtech.cn</a> 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" <<a href="mailto:liuzq2002@126.com">liuzq2002@126.com</a>> =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""> =D4=DA2009-05-28=A3=AC"Zhiqiang Liu" &lt;liuzq2002@= <a href="http://126.com/" target="_blank">126.com</a>&gt; =D0=B4=B5=C0=A3=BA <BLOCKQUOTE id=3D"isReplyContent" style= =3D"PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px sol= id"><div> 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> 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; 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. Thanks for your suggestion. I will try to analyse the&nbsp;traffic&nbsp;= using&nbsp;wireshark. I would like to keep in touch with you.&nbsp;When we get&nbsp;any progre= ss, I will contact you. &nbsp; Best regards, Zhiqiang Liu<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>  oter"/><hr/> <a href=3D"<a href="http://512.mail.163.com/mailstamp/stamp/dz/activity.do?from=3Dfo=" target="_blank">http://512.mail.163.com/mailstamp/stamp/dz/activity.do?from=3Dfo=</a> 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> ------=_Part_17596_27235653.1243734463069-- ------------------------------ Message: 3 Date: Sun, 31 May 2009 10:58:31 +0530 From: sudarshan bisht <<a href="mailto:bisht.sudarshan@gmail.com">bisht.sudarshan@gmail.com</a>> Subject: Re: [gst-devel] How to save a stream from a network into a file To: Discussion of the development of GStreamer <<a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a>> Message-ID: <<a href="mailto:785339900905302228n1000adf8pcfd2aec674bf03cb@mail.gmail.com">785339900905302228n1000adf8pcfd2aec674bf03cb@mail.gmail.com</a>> 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 <<a href="mailto:zelalems@hotmail.com">zelalems@hotmail.com</a>>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!<<a href="http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx&mkt=en-us" target="_blank">http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx&mkt=en-us</a>> > > > ------------------------------------------------------------------------------ > 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. <a href="http://p.sf.net/sfu/creativitycat-com" target="_blank">http://p.sf.net/sfu/creativitycat-com</a> > _______________________________________________ > gstreamer-devel mailing list > <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> > <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a> > > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... ------------------------------ Message: 4 Date: Sun, 31 May 2009 11:18:51 +0530 From: sudarshan bisht <<a href="mailto:bisht.sudarshan@gmail.com">bisht.sudarshan@gmail.com</a>> Subject: Re: [gst-devel] PLAy->PAUSE Issue with alsasink To: Discussion of the development of GStreamer <<a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a>> Message-ID: <<a href="mailto:785339900905302248x33561748ve5dc658be3c7ac00@mail.gmail.com">785339900905302248x33561748ve5dc658be3c7ac00@mail.gmail.com</a>> 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 <<a href="mailto:sikkim.suresh@gmail.com">sikkim.suresh@gmail.com</a>>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. <a href="http://p.sf.net/sfu/creativitycat-com" target="_blank">http://p.sf.net/sfu/creativitycat-com</a> > _______________________________________________ > gstreamer-devel mailing list > <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> > <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a> > > -- 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. <a href="http://p.sf.net/sfu/creativitycat-com" target="_blank">http://p.sf.net/sfu/creativitycat-com</a> ------------------------------ _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a> End of gstreamer-devel Digest, Vol 36, Issue 94 *********************************************** </blockquote></div>