From ruben.medellin.c at gmail.com Thu Jan 1 03:03:43 2009 From: ruben.medellin.c at gmail.com (=?ISO-8859-1?Q?Rub=E9n_Medell=EDn?=) Date: Wed, 31 Dec 2008 20:03:43 -0600 Subject: [gst-devel] Getting an EOS signal on tcpserversrc Message-ID: Hi. I am developing an application using pygst and tcpserversrc with tcpclientsink. Aside some unnecessary stuff for that matter, my program is as the one in this tutorial [ http://www.jejik.com/articles/2007/01/streaming_audio_over_tcp_with_python-gstreamer/], adding a bus_watch on each side (server and client) that listens for an EOS message. I am aware that network streaming is used principally for live stream, but in this case I want to be able to detect when a stream sent to the server has ended. Tcpserversrc does not provide an EOS message after is not receiving data, and I don't know if that is the functionality or is a bug. In the debug file I can see that when the audio in the server, which goes through an alsasink, has stopped playing, I get the following message: basesrc gstbasesrc.c:2056:gst_base_src_loop: pausing after gst_base_src_get_range() = wrong-state but I am unable to find the reason for that error. The pipeline doesn't go to NULL state neither. Using gstreamer 0.10.18 with pygst 0.10.11 The same message can be obtained using the command lines at the server and client sides, respectively gst-launch-0.10 tcpserversrc host=localhost port=8080 ! decodebin ! audioconvert ! alsasink gst-launch-0.10 filesrc location=/home/test.mp3 ! tcpclientsink host=localhost port=8080 The full debug file for the server is here [http://pastie.org/349940], with the mentioned line highlighted. Thanks in Advance Ruben M -------------- next part -------------- An HTML attachment was scrubbed... URL: From vaisakh.n at nestgroup.net Fri Jan 2 04:26:16 2009 From: vaisakh.n at nestgroup.net (Vaisakh N.) Date: Fri, 2 Jan 2009 08:56:16 +0530 Subject: [gst-devel] Hi Message-ID: <245245F712CBF74D865D90E99497849F780B3F@mail-3.tvm.nestgroup.net> I installed gstreamer 0.10.21.and installed gst-ugly plugin -0.10.10 .... and tried to run > ./gst-launch -v filesrc location=/usr/share/sounds/KDE_Logout_1.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! alsasink then its shows error like this /************************************************/ Setting pipeline to PAUSED ... Pipeline is PREROLLING ... /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:sink: caps = audio/x-vorbis /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:src: caps = audio/x-raw-float, rate=(int)44100, channels=(int)2, endianness=(int)1234, width=(int)32 /**************************************/ what to do? please help me!! advance thanks.. ***** Confidentiality Statement/Disclaimer ***** This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bisht.sudarshan at gmail.com Fri Jan 2 07:14:13 2009 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Fri, 2 Jan 2009 11:44:13 +0530 Subject: [gst-devel] Hi In-Reply-To: <245245F712CBF74D865D90E99497849F780B3F@mail-3.tvm.nestgroup.net> References: <245245F712CBF74D865D90E99497849F780B3F@mail-3.tvm.nestgroup.net> Message-ID: <785339900901012214i48f38416we241972b620e363@mail.gmail.com> Hi , Why are u using audioconvert and audioresample ? is there any specific reason ? On Fri, Jan 2, 2009 at 8:56 AM, Vaisakh N. wrote: > > I installed gstreamer 0.10.21.and installed gst-ugly plugin -0.10.10 .... > and tried to run > > > ./gst-launch -v filesrc location=/usr/share/sounds/KDE_Logout_1.ogg ! > oggdemux ! vorbisdec ! audioconvert ! audioresample ! alsasink > > > > then > its shows error like this > > /************************************************/ > > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:sink: caps = > audio/x-vorbis > /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:src: caps = > audio/x-raw-float, rate=(int)44100, channels=(int)2, endianness=(int)1234, > width=(int)32 > > /**************************************/ > what to do? > please help me!! > advance thanks.. > > ***** Confidentiality Statement/Disclaimer ***** > > This message and any attachments is intended for the sole use of the > intended recipient. It may contain confidential information. Any > unauthorized use, dissemination or modification is strictly prohibited. If > you are not the intended recipient, please notify the sender immediately > then delete it from all your systems, and do not copy, use or print. > Internet communications are not secure and it is the responsibility of the > recipient to make sure that it is virus/malicious code exempt. > The company/sender cannot be responsible for any unauthorized alterations > or modifications made to the contents. If you require any form of > confirmation of the contents, please contact the company/sender. The > company/sender is not liable for any errors or omissions in the content of > this message. > > > ------------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From vaisakh.n at nestgroup.net Fri Jan 2 07:47:46 2009 From: vaisakh.n at nestgroup.net (Vaisakh N.) Date: Fri, 2 Jan 2009 12:17:46 +0530 Subject: [gst-devel] Hi Message-ID: <245245F712CBF74D865D90E99497849F780B40@mail-3.tvm.nestgroup.net> @sudarshan bisht Hi, If I didnt used that audioconvert ! audioresample ! its shows that there is no link between vorbisdec and alsasink...... thats why I put that. its actually works for gst-launch-i386 (in /usr/lib ) -pre installed during os installation ... but its not working with gstreamer -0.10.21. ( in my local directory) just now installed.. ***** Confidentiality Statement/Disclaimer ***** This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bisht.sudarshan at gmail.com Fri Jan 2 08:02:00 2009 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Fri, 2 Jan 2009 12:32:00 +0530 Subject: [gst-devel] Hi In-Reply-To: <245245F712CBF74D865D90E99497849F780B40@mail-3.tvm.nestgroup.net> References: <245245F712CBF74D865D90E99497849F780B40@mail-3.tvm.nestgroup.net> Message-ID: <785339900901012302o19d98dbbp512345ba77ab0586@mail.gmail.com> ok ,, do gst-inspect vorbisdec and see the src pad caps of it , does it matches with sink caps of alsasink ? On Fri, Jan 2, 2009 at 12:17 PM, Vaisakh N. wrote: > > @sudarshan bisht > Hi, > > If I didnt used that audioconvert ! audioresample ! > its shows that there is no link between vorbisdec and alsasink...... > > thats why I put that. > > its actually works for gst-launch-i386 (in /usr/lib ) -pre installed during > os installation ... > > but its not working with gstreamer -0.10.21. ( in my local directory) > just now installed.. > > ***** Confidentiality Statement/Disclaimer ***** > > This message and any attachments is intended for the sole use of the > intended recipient. It may contain confidential information. Any > unauthorized use, dissemination or modification is strictly prohibited. If > you are not the intended recipient, please notify the sender immediately > then delete it from all your systems, and do not copy, use or print. > Internet communications are not secure and it is the responsibility of the > recipient to make sure that it is virus/malicious code exempt. > The company/sender cannot be responsible for any unauthorized alterations > or modifications made to the contents. If you require any form of > confirmation of the contents, please contact the company/sender. The > company/sender is not liable for any errors or omissions in the content of > this message. > > > ------------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From kumarkm at tataelxsi.co.in Fri Jan 2 08:08:34 2009 From: kumarkm at tataelxsi.co.in (Kumar) Date: Fri, 2 Jan 2009 12:38:34 +0530 Subject: [gst-devel] G711 & G726 payload packetizer merging Message-ID: <003e01c96ca8$edb816e0$60033c0a@telxsi.com> Hi all, The G711 & G726 RTP payload packetizers are present in different gst-good-plugin packages. ( 0.10.2 & 0.10.10). Currently I'm merging G711 from Good 0.10.2 to Good 0.10.10. After making logical changes in Makefile.in & Makefile.am in gst/rtp directory of Good 0.10.10 plug-in package, I'm getting the following compilation error: libtool: link: `libgstrtp_la-gstrtpg711pay.lo' is not a valid libtool obje can anyone tell, why this error is coming? Is there any gst-good plug-in package which both G711 & G726 RTP payload packetizers are present? Thanks in advance.. Regards, Kumar KM The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From bisht.sudarshan at gmail.com Fri Jan 2 08:13:13 2009 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Fri, 2 Jan 2009 12:43:13 +0530 Subject: [gst-devel] RTP C Examples In-Reply-To: References: Message-ID: <785339900901012313w6cd1f578rfbfe7656bede0b8a@mail.gmail.com> Hi ,, Acutally audioresample and rtpamrpay elements can not be linked since both have differenct caps . So better try some other combination. On Wed, Dec 31, 2008 at 9:33 PM, Jack Foster wrote: > Hi gang! > > I see lots of examples that have CLI to gStreamer but surely people are > building applications with C. It seems like an awesome framework but I'm > just trying to get my head around it. I would like to see more C examples. > > > In any case, I'm trying to a simple audio rtp sender/receiver working but > am having a hard time translating the example from the gstrtpbin > documentation to C. I'll then post a working copy on the mailing list for > others running into the same issue. > > I really have no idea if the errors are related to the mad plugin or the > pads not linked together properly. > > Attached is the error and the program. Any help would be greatly > appreciated. > > Thanks! > Jack > > > ------------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From dhanraj.b at allaboutif.com Fri Jan 2 10:13:48 2009 From: dhanraj.b at allaboutif.com (Dhanraj Birader) Date: Fri, 2 Jan 2009 14:43:48 +0530 Subject: [gst-devel] udpsrc on davinci board Message-ID: <9536d8fe0901020113u133c5fe1h65a8b102633913a8@mail.gmail.com> Hi, why udpsrc plugin is not working on davinci board. regards dhanraj -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilboed at gmail.com Fri Jan 2 10:30:01 2009 From: bilboed at gmail.com (Edward Hervey) Date: Fri, 02 Jan 2009 10:30:01 +0100 Subject: [gst-devel] udpsrc on davinci board In-Reply-To: <9536d8fe0901020113u133c5fe1h65a8b102633913a8@mail.gmail.com> References: <9536d8fe0901020113u133c5fe1h65a8b102633913a8@mail.gmail.com> Message-ID: <1230888601.13816.1.camel@joder> On Fri, 2009-01-02 at 14:43 +0530, Dhanraj Birader wrote: > Hi, > > why udpsrc plugin is not working on davinci board. Because you haven't switched the board on ? Seriously though, you will have to give us a *lot* more information for us to give you a sensible answer. Like, gstreamer-related debug information. Edward > > > > > > > > > > > regards > dhanraj > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From marco at teknusi.org Fri Jan 2 10:28:35 2009 From: marco at teknusi.org (marco) Date: Fri, 2 Jan 2009 10:28:35 +0100 Subject: [gst-devel] Adder element and pipeline rebuilding Message-ID: <5FF0561C-D5AB-4116-9BB4-7AC4045035D9@teknusi.org> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, i am trying to have this setup working: i have a pipeline with an adder element placed just after a decodebin element. what i want to do is adding a pad to the adder element while the pipeline is in GST_STATE_PLAYING. so i create a new bin for the new source, i add it to the pipeline, then i should link it to the adder. which is the right sequence to do this? i have problems and crash when i try to do the linking, the pipeline stalls and never restart. (it seems there is a lack of documentation about this family of problems) thanks in advance, marco -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (Darwin) iEYEARECAAYFAkld3kMACgkQ5A1A6ak/SoVibACeNYAy3iX8ifir2D2nUZmhjPm2 G9YAn1XTwzfyuvsWUlGkdAyvU0vUQsJ2 =0aGb -----END PGP SIGNATURE----- From bilboed at gmail.com Fri Jan 2 10:32:21 2009 From: bilboed at gmail.com (Edward Hervey) Date: Fri, 02 Jan 2009 10:32:21 +0100 Subject: [gst-devel] Hi In-Reply-To: <785339900901012302o19d98dbbp512345ba77ab0586@mail.gmail.com> References: <245245F712CBF74D865D90E99497849F780B40@mail-3.tvm.nestgroup.net> <785339900901012302o19d98dbbp512345ba77ab0586@mail.gmail.com> Message-ID: <1230888741.13816.3.camel@joder> And here's the first 2009 little extra hint: To convert between float raw audio (audio/x-raw-float) and integer raw audio (audio/x-raw-int) you need to use "audioconvert". Edward On Fri, 2009-01-02 at 12:32 +0530, sudarshan bisht wrote: > ok ,, do gst-inspect vorbisdec and see the src pad caps of it , does > it matches with sink caps of alsasink ? > > On Fri, Jan 2, 2009 at 12:17 PM, Vaisakh N. > wrote: > > @sudarshan bisht > Hi, > > If I didnt used that audioconvert ! audioresample ! > its shows that there is no link between vorbisdec and > alsasink...... > > thats why I put that. > > its actually works for gst-launch-i386 (in /usr/lib ) -pre > installed during > os installation ... > > but its not working with gstreamer -0.10.21. ( in my > local directory) just now installed.. > > > > ***** Confidentiality Statement/Disclaimer ***** > > This message and any attachments is intended for the sole use > of the intended recipient. It may contain confidential > information. Any unauthorized use, dissemination or > modification is strictly prohibited. If you are not the > intended recipient, please notify the sender immediately then > delete it from all your systems, and do not copy, use or > print. Internet communications are not secure and it is the > responsibility of the recipient to make sure that it is > virus/malicious code exempt. > The company/sender cannot be responsible for any unauthorized > alterations or modifications made to the contents. If you > require any form of confirmation of the contents, please > contact the company/sender. The company/sender is not liable > for any errors or omissions in the content of this message. > > ------------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > -- > Regards, > > Sudarshan Bisht > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From bisht.sudarshan at gmail.com Fri Jan 2 10:33:15 2009 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Fri, 2 Jan 2009 15:03:15 +0530 Subject: [gst-devel] Adder element and pipeline rebuilding In-Reply-To: <5FF0561C-D5AB-4116-9BB4-7AC4045035D9@teknusi.org> References: <5FF0561C-D5AB-4116-9BB4-7AC4045035D9@teknusi.org> Message-ID: <785339900901020133h5d198c67kc07eb52572941b20@mail.gmail.com> HI , First set your pipeline to GST_STATE_PAUSED state then add the new element and then again set pipeline to GST_STATE_PLAYING state. On Fri, Jan 2, 2009 at 2:58 PM, marco wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hi, > > i am trying to have this setup working: > i have a pipeline with an adder element placed just after a decodebin > element. > what i want to do is adding a pad to the adder element while the > pipeline is in GST_STATE_PLAYING. > so i create a new bin for the new source, i add it to the pipeline, > then i should link it to the adder. > which is the right sequence to do this? > i have problems and crash when i try to do the linking, the pipeline > stalls and never restart. > > (it seems there is a lack of documentation about this family of > problems) > > thanks in advance, > > marco > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.9 (Darwin) > > iEYEARECAAYFAkld3kMACgkQ5A1A6ak/SoVibACeNYAy3iX8ifir2D2nUZmhjPm2 > G9YAn1XTwzfyuvsWUlGkdAyvU0vUQsJ2 > =0aGb > -----END PGP SIGNATURE----- > > > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From marco at teknusi.org Fri Jan 2 10:46:25 2009 From: marco at teknusi.org (marco) Date: Fri, 2 Jan 2009 10:46:25 +0100 Subject: [gst-devel] Adder element and pipeline rebuilding In-Reply-To: <785339900901020133h5d198c67kc07eb52572941b20@mail.gmail.com> References: <5FF0561C-D5AB-4116-9BB4-7AC4045035D9@teknusi.org> <785339900901020133h5d198c67kc07eb52572941b20@mail.gmail.com> Message-ID: <457DF49F-451E-4BB5-9A2D-122AC5160F25@teknusi.org> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Il giorno 02/gen/09, alle ore 10:33, sudarshan bisht ha scritto: > HI , > First set your pipeline to GST_STATE_PAUSED state then add > the new element and then again set pipeline to GST_STATE_PLAYING > state. thanks for your fast answer. it seems it does not work, it stalls or it gives errors in the flow. the following is the code i am executing: void add_sine(){ GstElement *tsine; tsine = get_silence_bin(); gst_element_set_state(pipeline, GST_STATE_PAUSED); gst_bin_add(GST_BIN(pipeline), tsine); gst_pad_link(gst_element_get_static_pad(tsine, "src"), gst_element_get_request_pad(adder, "sink1")); gst_element_set_state(pipeline, GST_STATE_PLAYING); } this brings to a flow error. if i add a : gst_element_set_state(tsine, GST_STATE_PAUSED); before adding it to the pipeline, it simply stalls (i never get an async-done from the pipeline, that should be the message posted to the bus when the pipeline is in playing state again. just for your information: this pipeline has a tee element outputting to two different sink elements (alsasink and shout2send); any hint on how to get this working? marco -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (Darwin) iEYEARECAAYFAkld4nEACgkQ5A1A6ak/SoXO2QCgolmRiZRm8GSEeRnQalLBOrFT lDgAnRL5iT7LTYkYFfAl7IBOGe2uji3G =mkNJ -----END PGP SIGNATURE----- From bisht.sudarshan at gmail.com Fri Jan 2 10:52:00 2009 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Fri, 2 Jan 2009 15:22:00 +0530 Subject: [gst-devel] Adder element and pipeline rebuilding In-Reply-To: <457DF49F-451E-4BB5-9A2D-122AC5160F25@teknusi.org> References: <5FF0561C-D5AB-4116-9BB4-7AC4045035D9@teknusi.org> <785339900901020133h5d198c67kc07eb52572941b20@mail.gmail.com> <457DF49F-451E-4BB5-9A2D-122AC5160F25@teknusi.org> Message-ID: <785339900901020152i5dd087a4of13c46c74dd75aec@mail.gmail.com> Hi ,, Add the tsine element before , link it in later in GST_STATE_PAUSED state. On Fri, Jan 2, 2009 at 3:16 PM, marco wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > > Il giorno 02/gen/09, alle ore 10:33, sudarshan bisht ha scritto: > > > HI , > > First set your pipeline to GST_STATE_PAUSED state then add > > the new element and then again set pipeline to GST_STATE_PLAYING > > state. > > thanks for your fast answer. > it seems it does not work, it stalls or it gives errors in the flow. > the following is the code i am executing: > > void add_sine(){ > GstElement *tsine; > > tsine = get_silence_bin(); > gst_element_set_state(pipeline, GST_STATE_PAUSED); > > gst_bin_add(GST_BIN(pipeline), tsine); > > > gst_pad_link(gst_element_get_static_pad(tsine, "src"), > gst_element_get_request_pad(adder, "sink1")); > > gst_element_set_state(pipeline, GST_STATE_PLAYING); > } > > this brings to a flow error. if i add a : > gst_element_set_state(tsine, GST_STATE_PAUSED); > before adding it to the pipeline, it simply stalls (i never get an > async-done from the pipeline, that should be the message posted to the > bus when the pipeline is in playing state again. > just for your information: this pipeline has a tee element outputting > to two different sink elements (alsasink and shout2send); > any hint on how to get this working? > > marco > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.9 (Darwin) > > iEYEARECAAYFAkld4nEACgkQ5A1A6ak/SoXO2QCgolmRiZRm8GSEeRnQalLBOrFT > lDgAnRL5iT7LTYkYFfAl7IBOGe2uji3G > =mkNJ > -----END PGP SIGNATURE----- > > > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From marco at teknusi.org Fri Jan 2 11:11:56 2009 From: marco at teknusi.org (marco) Date: Fri, 2 Jan 2009 11:11:56 +0100 Subject: [gst-devel] Adder element and pipeline rebuilding In-Reply-To: <785339900901020152i5dd087a4of13c46c74dd75aec@mail.gmail.com> References: <5FF0561C-D5AB-4116-9BB4-7AC4045035D9@teknusi.org> <785339900901020133h5d198c67kc07eb52572941b20@mail.gmail.com> <457DF49F-451E-4BB5-9A2D-122AC5160F25@teknusi.org> <785339900901020152i5dd087a4of13c46c74dd75aec@mail.gmail.com> Message-ID: -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Il giorno 02/gen/09, alle ore 10:52, sudarshan bisht ha scritto: > Hi ,, > Add the tsine element before , link it in later in > GST_STATE_PAUSED state. in this way i have no errors but the pipeline stalls and the alsa output sink does not play nothing. i suspect it could be the shout2send sink not restarting... i have no idea of what happens : ( marco -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (Darwin) iEYEARECAAYFAkld6GwACgkQ5A1A6ak/SoXcXQCgoicsOlVVyObHatS5xTvoQzlX IcgAn14ZCDynRAnENrTB38JmnmO6u+OW =wrQE -----END PGP SIGNATURE----- From ensonic at hora-obscura.de Fri Jan 2 11:59:39 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Fri, 02 Jan 2009 12:59:39 +0200 Subject: [gst-devel] Hi In-Reply-To: <245245F712CBF74D865D90E99497849F780B3F@mail-3.tvm.nestgroup.net> References: <245245F712CBF74D865D90E99497849F780B3F@mail-3.tvm.nestgroup.net> Message-ID: <495DF39B.6000106@hora-obscura.de> Vaisakh N. schrieb: > > I installed gstreamer 0.10.21.and installed gst-ugly plugin -0.10.10 .... Do you also have gst-plugins-base and gst-plugins-good installed? > and tried to run > >> ./gst-launch -v filesrc location=/usr/share/sounds/KDE_Logout_1.ogg ! > oggdemux ! vorbisdec ! audioconvert ! audioresample ! alsasink > > > > then > its shows error like this > > /************************************************/ > > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:sink: caps = > audio/x-vorbis > /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:src: caps = > audio/x-raw-float, rate=(int)44100, channels=(int)2, > endianness=(int)1234, width=(int)32 Thats all? is there no more output following? Stefan > > /**************************************/ > what to do? > please help me!! > advance thanks.. > > ***** Confidentiality Statement/Disclaimer ***** > > This message and any attachments is intended for the sole use of the > intended recipient. It may contain confidential information. Any > unauthorized use, dissemination or modification is strictly prohibited. > If you are not the intended recipient, please notify the sender > immediately then delete it from all your systems, and do not copy, use > or print. Internet communications are not secure and it is the > responsibility of the recipient to make sure that it is virus/malicious > code exempt. > The company/sender cannot be responsible for any unauthorized > alterations or modifications made to the contents. If you require any > form of confirmation of the contents, please contact the company/sender. > The company/sender is not liable for any errors or omissions in the > content of this message. > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------------ > > > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From ensonic at hora-obscura.de Fri Jan 2 12:02:08 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Fri, 02 Jan 2009 13:02:08 +0200 Subject: [gst-devel] Adder element and pipeline rebuilding In-Reply-To: <5FF0561C-D5AB-4116-9BB4-7AC4045035D9@teknusi.org> References: <5FF0561C-D5AB-4116-9BB4-7AC4045035D9@teknusi.org> Message-ID: <495DF430.1090704@hora-obscura.de> marco schrieb: > Hi, > > i am trying to have this setup working: > i have a pipeline with an adder element placed just after a decodebin > element. > what i want to do is adding a pad to the adder element while the > pipeline is in GST_STATE_PLAYING. > so i create a new bin for the new source, i add it to the pipeline, > then i should link it to the adder. > which is the right sequence to do this? > i have problems and crash when i try to do the linking, the pipeline > stalls and never restart. > > (it seems there is a lack of documentation about this family of > problems) Have a look at gstreamer/docs/design/part-block.txt Stefan > > thanks in advance, > > marco ------------------------------------------------------------------------------ _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From vaisakh.n at nestgroup.net Fri Jan 2 12:04:14 2009 From: vaisakh.n at nestgroup.net (vaisakh.n) Date: Fri, 02 Jan 2009 16:34:14 +0530 Subject: [gst-devel] Hi In-Reply-To: <495DF39B.6000106@hora-obscura.de> References: <245245F712CBF74D865D90E99497849F780B3F@mail-3.tvm.nestgroup.net> <495DF39B.6000106@hora-obscura.de> Message-ID: <495DF4AE.40704@nestgroup.net> Stefan Kost wrote: >Vaisakh N. schrieb: > > >>I installed gstreamer 0.10.21.and installed gst-ugly plugin -0.10.10 .... >> >> > >Do you also have gst-plugins-base and gst-plugins-good installed? > > > >>and tried to run >> >> >> >>>./gst-launch -v filesrc location=/usr/share/sounds/KDE_Logout_1.ogg ! >>> >>> >>oggdemux ! vorbisdec ! audioconvert ! audioresample ! alsasink >> >> >> >>then >>its shows error like this >> >>/************************************************/ >> >>Setting pipeline to PAUSED ... >>Pipeline is PREROLLING ... >>/GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:sink: caps = >>audio/x-vorbis >>/GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:src: caps = >>audio/x-raw-float, rate=(int)44100, channels=(int)2, >>endianness=(int)1234, width=(int)32 >> >> > >Thats all? is there no more output following? > >Stefan > > > > >>/**************************************/ >>what to do? >>please help me!! >>advance thanks.. >> >>***** Confidentiality Statement/Disclaimer ***** >> >>This message and any attachments is intended for the sole use of the >>intended recipient. It may contain confidential information. Any >>unauthorized use, dissemination or modification is strictly prohibited. >>If you are not the intended recipient, please notify the sender >>immediately then delete it from all your systems, and do not copy, use >>or print. Internet communications are not secure and it is the >>responsibility of the recipient to make sure that it is virus/malicious >>code exempt. >>The company/sender cannot be responsible for any unauthorized >>alterations or modifications made to the contents. If you require any >>form of confirmation of the contents, please contact the company/sender. >>The company/sender is not liable for any errors or omissions in the >>content of this message. >> >> >>------------------------------------------------------------------------ >> >>------------------------------------------------------------------------------ >> >> >>------------------------------------------------------------------------ >> >>_______________________________________________ >>gstreamer-devel mailing list >>gstreamer-devel at lists.sourceforge.net >>https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > >------------------------------------------------------------------------------ >_______________________________________________ >gstreamer-devel mailing list >gstreamer-devel at lists.sourceforge.net >https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > no its getting struck ***** Confidentiality Statement/Disclaimer ***** This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ensonic at hora-obscura.de Fri Jan 2 12:14:58 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Fri, 02 Jan 2009 13:14:58 +0200 Subject: [gst-devel] Hi In-Reply-To: <495DF4AE.40704@nestgroup.net> References: <245245F712CBF74D865D90E99497849F780B3F@mail-3.tvm.nestgroup.net> <495DF39B.6000106@hora-obscura.de> <495DF4AE.40704@nestgroup.net> Message-ID: <495DF732.3030508@hora-obscura.de> vaisakh.n schrieb: > Stefan Kost wrote: >> Vaisakh N. schrieb: >> >>> I installed gstreamer 0.10.21.and installed gst-ugly plugin -0.10.10 .... >>> >> >> Do you also have gst-plugins-base and gst-plugins-good installed? >> >> >>> and tried to run >>> >>> >>>> ./gst-launch -v filesrc location=/usr/share/sounds/KDE_Logout_1.ogg ! >>>> >>> oggdemux ! vorbisdec ! audioconvert ! audioresample ! alsasink >>> >>> >>> >>> then >>> its shows error like this >>> >>> /************************************************/ >>> >>> Setting pipeline to PAUSED ... >>> Pipeline is PREROLLING ... >>> /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:sink: caps = >>> audio/x-vorbis >>> /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:src: caps = >>> audio/x-raw-float, rate=(int)44100, channels=(int)2, >>> endianness=(int)1234, width=(int)32 >>> >> >> Thats all? is there no more output following? >> >> Stefan >> >> >> >>> /**************************************/ >>> what to do? >>> please help me!! >>> advance thanks.. >>> > no its getting struck play totaly fine here. Do you run gstreamer uninstalled (noticed the ./gst-launch)? Try gdb and get us a backtrace where its stuck or do GST_DEBUG="*:4" gst-launch 2>debug.log filesrc ... and send us the gzipped log. Stefan gst-launch -v filesrc location=/opt/kde3/share/sounds/KDE_Logout_1.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! alsasink Setting pipeline to PAUSED ... Pipeline is PREROLLING ... /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:sink: caps = audio/x-vorbis /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:src: caps = audio/x-raw-float, rate=(int)44100, channels=(int)2, endianness=(int)1234, width=(int)32 /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = audio/x-raw-float, rate=(int)44100, channels=(int)2, endianness=(int)1234, width=(int)32 /GstPipeline:pipeline0/GstSpeexResample:speexresample0.GstPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 /GstPipeline:pipeline0/GstSpeexResample:speexresample0.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 /GstPipeline:pipeline0/GstAlsaSink:alsasink0.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstAudioSinkClock ^CCaught interrupt -- handling interrupt. Interrupt: Stopping pipeline ... Execution ended after 4904080068 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... /GstPipeline:pipeline0/GstAlsaSink:alsasink0.GstPad:sink: caps = NULL /GstPipeline:pipeline0/GstSpeexResample:speexresample0.GstPad:src: caps = NULL /GstPipeline:pipeline0/GstSpeexResample:speexresample0.GstPad:sink: caps = NULL /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps = NULL /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = NULL /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:src: caps = NULL /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:sink: caps = NULL /GstPipeline:pipeline0/GstOggDemux:oggdemux0.GstOggPad:serial_7067c2bf: caps = NULL Setting pipeline to NULL ... FREEING pipeline ... From marco at teknusi.org Fri Jan 2 12:40:11 2009 From: marco at teknusi.org (marco) Date: Fri, 2 Jan 2009 12:40:11 +0100 Subject: [gst-devel] Adder element and pipeline rebuilding In-Reply-To: <495DF430.1090704@hora-obscura.de> References: <5FF0561C-D5AB-4116-9BB4-7AC4045035D9@teknusi.org> <495DF430.1090704@hora-obscura.de> Message-ID: <667753AD-73C0-45CB-81A8-CD51F3008617@teknusi.org> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Il giorno 02/gen/09, alle ore 12:02, Stefan Kost ha scritto: > Have a look at gstreamer/docs/design/part-block.txt had a look and implemented some code, i can get to block a pad only when an EOS is reached, that seems strange to me...or this is the normal behaviour? marco -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (Darwin) iEYEARECAAYFAkld/RsACgkQ5A1A6ak/SoVoXgCgiwnQnHH44nQcKbKRoa5nmRTb 3fEAnRbs6bgIYCrPZZAmRvgg/u/u6zG7 =46u4 -----END PGP SIGNATURE----- From wim.taymans at gmail.com Fri Jan 2 13:21:07 2009 From: wim.taymans at gmail.com (Wim Taymans) Date: Fri, 2 Jan 2009 13:21:07 +0100 Subject: [gst-devel] RTP C Examples In-Reply-To: <785339900901012313w6cd1f578rfbfe7656bede0b8a@mail.gmail.com> References: <785339900901012313w6cd1f578rfbfe7656bede0b8a@mail.gmail.com> Message-ID: gst_element_link_many() will not automatically link the payloader to the rtpbin, you need to manually request a sink pad first (like you did) and link it to the srcpad of the payloader. Wim On Fri, Jan 2, 2009 at 8:13 AM, sudarshan bisht wrote: > Hi ,, > Acutally audioresample and rtpamrpay elements can not be linked > since both have differenct caps . So better try some other combination. > > On Wed, Dec 31, 2008 at 9:33 PM, Jack Foster > wrote: >> >> Hi gang! >> >> I see lots of examples that have CLI to gStreamer but surely people are >> building applications with C. It seems like an awesome framework but I'm >> just trying to get my head around it. I would like to see more C examples. >> >> In any case, I'm trying to a simple audio rtp sender/receiver working but >> am having a hard time translating the example from the gstrtpbin >> documentation to C. I'll then post a working copy on the mailing list for >> others running into the same issue. >> >> I really have no idea if the errors are related to the mad plugin or the >> pads not linked together properly. >> >> Attached is the error and the program. Any help would be greatly >> appreciated. >> >> Thanks! >> Jack >> >> >> ------------------------------------------------------------------------------ >> >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > > > -- > Regards, > > Sudarshan Bisht > > ------------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From wim.taymans at gmail.com Fri Jan 2 13:22:42 2009 From: wim.taymans at gmail.com (Wim Taymans) Date: Fri, 2 Jan 2009 13:22:42 +0100 Subject: [gst-devel] RTP C Examples In-Reply-To: References: <785339900901012313w6cd1f578rfbfe7656bede0b8a@mail.gmail.com> Message-ID: Also, you need an amr encoder before the amr payloader. Wim On Fri, Jan 2, 2009 at 1:21 PM, Wim Taymans wrote: > gst_element_link_many() will not automatically link the payloader to > the rtpbin, you need to manually request > a sink pad first (like you did) and link it to the srcpad of the payloader. > > Wim > > On Fri, Jan 2, 2009 at 8:13 AM, sudarshan bisht > wrote: >> Hi ,, >> Acutally audioresample and rtpamrpay elements can not be linked >> since both have differenct caps . So better try some other combination. >> >> On Wed, Dec 31, 2008 at 9:33 PM, Jack Foster >> wrote: >>> >>> Hi gang! >>> >>> I see lots of examples that have CLI to gStreamer but surely people are >>> building applications with C. It seems like an awesome framework but I'm >>> just trying to get my head around it. I would like to see more C examples. >>> >>> In any case, I'm trying to a simple audio rtp sender/receiver working but >>> am having a hard time translating the example from the gstrtpbin >>> documentation to C. I'll then post a working copy on the mailing list for >>> others running into the same issue. >>> >>> I really have no idea if the errors are related to the mad plugin or the >>> pads not linked together properly. >>> >>> Attached is the error and the program. Any help would be greatly >>> appreciated. >>> >>> Thanks! >>> Jack >>> >>> >>> ------------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >> >> >> >> -- >> Regards, >> >> Sudarshan Bisht >> >> ------------------------------------------------------------------------------ >> >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > From marco at teknusi.org Fri Jan 2 13:38:48 2009 From: marco at teknusi.org (marco) Date: Fri, 2 Jan 2009 13:38:48 +0100 Subject: [gst-devel] Adder element and pipeline rebuilding In-Reply-To: <667753AD-73C0-45CB-81A8-CD51F3008617@teknusi.org> References: <5FF0561C-D5AB-4116-9BB4-7AC4045035D9@teknusi.org> <495DF430.1090704@hora-obscura.de> <667753AD-73C0-45CB-81A8-CD51F3008617@teknusi.org> Message-ID: <5D753B7D-B268-462B-A362-6C9E860BC817@teknusi.org> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Il giorno 02/gen/09, alle ore 12:40, marco ha scritto: > had a look and implemented some code, i can get to block a pad only > when an EOS is reached, that seems strange to me...or this is the > normal behaviour? it seems that bloking an adder element sinkpad only succeeds when an EOS is reached by the source. i cannot understand if this is a bug or what. marco -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (Darwin) iEYEARECAAYFAkleCtgACgkQ5A1A6ak/SoXWlQCfZFRc4ZNdxRvIk+ixaIbVHYV+ crsAniuJsGxpDc5NsL3QmTzw2CePrDGL =gD2t -----END PGP SIGNATURE----- From wim.taymans at gmail.com Fri Jan 2 16:05:09 2009 From: wim.taymans at gmail.com (Wim Taymans) Date: Fri, 02 Jan 2009 16:05:09 +0100 Subject: [gst-devel] G711 & G726 payload packetizer merging In-Reply-To: <003e01c96ca8$edb816e0$60033c0a@telxsi.com> References: <003e01c96ca8$edb816e0$60033c0a@telxsi.com> Message-ID: <1230908709.31475.49.camel@metal> On Fri, 2009-01-02 at 12:38 +0530, Kumar wrote: > Hi all, > > The G711 & G726 RTP payload packetizers are present in different > gst-good-plugin packages. ( 0.10.2 & 0.10.10). > Currently I'm merging G711 from Good 0.10.2 to Good 0.10.10. After making > logical changes in Makefile.in & Makefile.am in gst/rtp directory of Good > 0.10.10 plug-in package, > I'm getting the following compilation error: > > libtool: link: `libgstrtp_la-gstrtpg711pay.lo' is not a valid libtool obje > > can anyone tell, why this error is coming? > Is there any gst-good plug-in package which both G711 & G726 RTP payload > packetizers are present? gst-plugins-good 0.10.10 has both elements. Wim > > Thanks in advance.. > > Regards, > Kumar KM > > > The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. > > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From wim.taymans at gmail.com Fri Jan 2 16:09:07 2009 From: wim.taymans at gmail.com (Wim Taymans) Date: Fri, 02 Jan 2009 16:09:07 +0100 Subject: [gst-devel] Adder element and pipeline rebuilding In-Reply-To: <5FF0561C-D5AB-4116-9BB4-7AC4045035D9@teknusi.org> References: <5FF0561C-D5AB-4116-9BB4-7AC4045035D9@teknusi.org> Message-ID: <1230908947.31475.53.camel@metal> On Fri, 2009-01-02 at 10:28 +0100, marco wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hi, > > i am trying to have this setup working: > i have a pipeline with an adder element placed just after a decodebin > element. > what i want to do is adding a pad to the adder element while the > pipeline is in GST_STATE_PLAYING. > so i create a new bin for the new source, i add it to the pipeline, > then i should link it to the adder. > which is the right sequence to do this? The sequence is: 1) leave the bin to READY 2) request a new pad from adder 3) link the requested pad to the bin 4) set the bin to PLAYING > i have problems and crash when i try to do the linking, the pipeline > stalls and never restart. Did you do gobject.threads_init() first? Wim > > (it seems there is a lack of documentation about this family of > problems) > > thanks in advance, > > marco > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.9 (Darwin) > > iEYEARECAAYFAkld3kMACgkQ5A1A6ak/SoVibACeNYAy3iX8ifir2D2nUZmhjPm2 > G9YAn1XTwzfyuvsWUlGkdAyvU0vUQsJ2 > =0aGb > -----END PGP SIGNATURE----- > > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From wim.taymans at gmail.com Fri Jan 2 16:22:39 2009 From: wim.taymans at gmail.com (Wim Taymans) Date: Fri, 02 Jan 2009 16:22:39 +0100 Subject: [gst-devel] RTP C Examples In-Reply-To: References: Message-ID: <1230909759.31475.58.camel@metal> On Wed, 2008-12-31 at 11:03 -0500, Jack Foster wrote: > Hi gang! > > I see lots of examples that have CLI to gStreamer but surely people > are building applications with C. It seems like an awesome framework > but I'm just trying to get my head around it. I would like to see > more C examples. > > In any case, I'm trying to a simple audio rtp sender/receiver working > but am having a hard time translating the example from the gstrtpbin > documentation to C. I'll then post a working copy on the mailing list > for others running into the same issue. > > I really have no idea if the errors are related to the mad plugin or > the pads not linked together properly. > > Attached is the error and the program. Any help would be greatly > appreciated. I added some C examples in CVS here: http://webcvs.freedesktop.org/gstreamer/gst-plugins-good/tests/examples/rtp/ Look at client-PCMA.c and server-alsasrc-PCMA.c Wim > > Thanks! > Jack > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From kumarkm at tataelxsi.co.in Fri Jan 2 16:26:54 2009 From: kumarkm at tataelxsi.co.in (Kumar) Date: Fri, 2 Jan 2009 20:56:54 +0530 Subject: [gst-devel] G711 & G726 payload packetizer merging In-Reply-To: <1230908709.31475.49.camel@metal> Message-ID: <000001c96cee$8b657c30$60033c0a@telxsi.com> Hi Wim, I could not find the source for the G711 RTP pay, but G726 payload packitizer is present in the gst-plugins-good 0.10.10 package. After installation of the package, gst-inspect will show only gstrtpg726 pay. can you pls give me more info on this. Thanks , Regards, Kumar KM -----Original Message----- From: Wim Taymans [mailto:wim.taymans at gmail.com] Sent: Friday, January 02, 2009 8:35 PM To: kumarkm at tataelxsi.co.in; Discussion of the development of GStreamer Subject: Re: [gst-devel] G711 & G726 payload packetizer merging On Fri, 2009-01-02 at 12:38 +0530, Kumar wrote: > Hi all, > > The G711 & G726 RTP payload packetizers are present in different > gst-good-plugin packages. ( 0.10.2 & 0.10.10). > Currently I'm merging G711 from Good 0.10.2 to Good 0.10.10. After making > logical changes in Makefile.in & Makefile.am in gst/rtp directory of Good > 0.10.10 plug-in package, > I'm getting the following compilation error: > > libtool: link: `libgstrtp_la-gstrtpg711pay.lo' is not a valid libtool obje > > can anyone tell, why this error is coming? > Is there any gst-good plug-in package which both G711 & G726 RTP payload > packetizers are present? gst-plugins-good 0.10.10 has both elements. Wim > > Thanks in advance.. > > Regards, > Kumar KM > > > The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. > > -------------------------------------------------------------------------- ---- > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From wim.taymans at gmail.com Fri Jan 2 16:29:33 2009 From: wim.taymans at gmail.com (Wim Taymans) Date: Fri, 02 Jan 2009 16:29:33 +0100 Subject: [gst-devel] G711 & G726 payload packetizer merging In-Reply-To: <000001c96cee$8b657c30$60033c0a@telxsi.com> References: <000001c96cee$8b657c30$60033c0a@telxsi.com> Message-ID: <1230910173.31475.62.camel@metal> On Fri, 2009-01-02 at 20:56 +0530, Kumar wrote: > Hi Wim, > > I could not find the source for the G711 RTP pay, but G726 payload > packitizer is present in the gst-plugins-good 0.10.10 package. > After installation of the package, gst-inspect will show only gstrtpg726 > pay. can you pls give me more info on this. rtppcma(de)pay/rtppcmu(de)pay are the G.711 a-law and mu-law (de)payloaders respectively. Wim > > Thanks , > > Regards, > Kumar KM > > -----Original Message----- > From: Wim Taymans [mailto:wim.taymans at gmail.com] > Sent: Friday, January 02, 2009 8:35 PM > To: kumarkm at tataelxsi.co.in; Discussion of the development of GStreamer > Subject: Re: [gst-devel] G711 & G726 payload packetizer merging > > > On Fri, 2009-01-02 at 12:38 +0530, Kumar wrote: > > Hi all, > > > > The G711 & G726 RTP payload packetizers are present in different > > gst-good-plugin packages. ( 0.10.2 & 0.10.10). > > Currently I'm merging G711 from Good 0.10.2 to Good 0.10.10. After making > > logical changes in Makefile.in & Makefile.am in gst/rtp directory of Good > > 0.10.10 plug-in package, > > I'm getting the following compilation error: > > > > libtool: link: `libgstrtp_la-gstrtpg711pay.lo' is not a valid libtool obje > > > > can anyone tell, why this error is coming? > > Is there any gst-good plug-in package which both G711 & G726 RTP payload > > packetizers are present? > > gst-plugins-good 0.10.10 has both elements. > > Wim > > > > > Thanks in advance.. > > > > Regards, > > Kumar KM > > > > > > The information contained in this electronic message and any attachments > to this message are intended for the exclusive use of the addressee(s) and > may contain proprietary, confidential or privileged information. If you are > not the intended recipient, you should not disseminate, distribute or copy > this e-mail. Please notify the sender immediately and destroy all copies of > this message and any attachments contained in it. > > > > -------------------------------------------------------------------------- > ---- > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From kumarkm at tataelxsi.co.in Fri Jan 2 16:54:17 2009 From: kumarkm at tataelxsi.co.in (Kumar) Date: Fri, 2 Jan 2009 21:24:17 +0530 Subject: [gst-devel] G711 & G726 payload packetizer merging In-Reply-To: <1230910173.31475.62.camel@metal> Message-ID: <000101c96cf2$5ef527a0$60033c0a@telxsi.com> Ya. got it. Thanks very much Wim. :) T=Regards, Kumar KM -----Original Message----- From: Wim Taymans [mailto:wim.taymans at gmail.com] Sent: Friday, January 02, 2009 9:00 PM To: kumarkm at tataelxsi.co.in Cc: 'Discussion of the development of GStreamer' Subject: RE: [gst-devel] G711 & G726 payload packetizer merging On Fri, 2009-01-02 at 20:56 +0530, Kumar wrote: > Hi Wim, > > I could not find the source for the G711 RTP pay, but G726 payload > packitizer is present in the gst-plugins-good 0.10.10 package. > After installation of the package, gst-inspect will show only gstrtpg726 > pay. can you pls give me more info on this. rtppcma(de)pay/rtppcmu(de)pay are the G.711 a-law and mu-law (de)payloaders respectively. Wim > > Thanks , > > Regards, > Kumar KM > > -----Original Message----- > From: Wim Taymans [mailto:wim.taymans at gmail.com] > Sent: Friday, January 02, 2009 8:35 PM > To: kumarkm at tataelxsi.co.in; Discussion of the development of GStreamer > Subject: Re: [gst-devel] G711 & G726 payload packetizer merging > > > On Fri, 2009-01-02 at 12:38 +0530, Kumar wrote: > > Hi all, > > > > The G711 & G726 RTP payload packetizers are present in different > > gst-good-plugin packages. ( 0.10.2 & 0.10.10). > > Currently I'm merging G711 from Good 0.10.2 to Good 0.10.10. After making > > logical changes in Makefile.in & Makefile.am in gst/rtp directory of Good > > 0.10.10 plug-in package, > > I'm getting the following compilation error: > > > > libtool: link: `libgstrtp_la-gstrtpg711pay.lo' is not a valid libtool obje > > > > can anyone tell, why this error is coming? > > Is there any gst-good plug-in package which both G711 & G726 RTP payload > > packetizers are present? > > gst-plugins-good 0.10.10 has both elements. > > Wim > > > > > Thanks in advance.. > > > > Regards, > > Kumar KM > > > > > > The information contained in this electronic message and any attachments > to this message are intended for the exclusive use of the addressee(s) and > may contain proprietary, confidential or privileged information. If you are > not the intended recipient, you should not disseminate, distribute or copy > this e-mail. Please notify the sender immediately and destroy all copies of > this message and any attachments contained in it. > > > > -------------------------------------------------------------------------- > ---- > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From jack.foster11 at gmail.com Fri Jan 2 17:00:13 2009 From: jack.foster11 at gmail.com (Jack Foster) Date: Fri, 2 Jan 2009 11:00:13 -0500 Subject: [gst-devel] RTP C Examples In-Reply-To: <1230909759.31475.58.camel@metal> References: <1230909759.31475.58.camel@metal> Message-ID: These are excellent! I'll try them out on Monday! Thanks a bunch, Jevin On Fri, Jan 2, 2009 at 10:22 AM, Wim Taymans wrote: > On Wed, 2008-12-31 at 11:03 -0500, Jack Foster wrote: > > Hi gang! > > > > I see lots of examples that have CLI to gStreamer but surely people > > are building applications with C. It seems like an awesome framework > > but I'm just trying to get my head around it. I would like to see > > more C examples. > > > > In any case, I'm trying to a simple audio rtp sender/receiver working > > but am having a hard time translating the example from the gstrtpbin > > documentation to C. I'll then post a working copy on the mailing list > > for others running into the same issue. > > > > I really have no idea if the errors are related to the mad plugin or > > the pads not linked together properly. > > > > Attached is the error and the program. Any help would be greatly > > appreciated. > > I added some C examples in CVS here: > > > http://webcvs.freedesktop.org/gstreamer/gst-plugins-good/tests/examples/rtp/ > > Look at client-PCMA.c and server-alsasrc-PCMA.c > > Wim > > > > > Thanks! > > Jack > > > ------------------------------------------------------------------------------ > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From marco at teknusi.org Fri Jan 2 19:17:35 2009 From: marco at teknusi.org (marco) Date: Fri, 2 Jan 2009 19:17:35 +0100 Subject: [gst-devel] Adder element and pipeline rebuilding In-Reply-To: <1230908947.31475.53.camel@metal> References: <5FF0561C-D5AB-4116-9BB4-7AC4045035D9@teknusi.org> <1230908947.31475.53.camel@metal> Message-ID: <9035D8D0-0C87-4B14-BCD6-1E3923F24AD1@teknusi.org> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Il giorno 02/gen/09, alle ore 16:09, Wim Taymans ha scritto: > The sequence is: > > 1) leave the bin to READY > 2) request a new pad from adder > 3) link the requested pad to the bin > 4) set the bin to PLAYING ok, tried this too and i did not succeed into having it to work. the code is: void add_sine(){ GstElement *tsine; int pstate; GstPad *newpad; GstPad *toblock; tsine = get_silence_bin(); gst_bin_add(GST_BIN(pipeline), tsine); //toblock = gst_ghost_pad_get_target( // gst_element_get_static_pad(adderbin, "sink0")); //gst_pad_set_blocked(toblock, TRUE); newpad = gst_ghost_pad_new("sink1", gst_element_get_request_pad(adder, "sink%d")); gst_element_add_pad(adderbin, newpad); gst_pad_link(gst_element_get_static_pad(tsine, "src"), gst_element_get_static_pad(adderbin, "sink1")); gst_element_set_state(tsine, GST_STATE_PLAYING); //gst_pad_set_blocked(toblock, FALSE); } >> i have problems and crash when i try to do the linking, the pipeline >> stalls and never restart. > > Did you do gobject.threads_init() first? yes, i have g_thread_init(NULL) call and gst_init() for first in the main (and i think that gst_init() does a conditional call to g_thread_init() when !g_thread_supported(), or am i wrong? anyway, thanks for your help, i am sorry to bother the list with so many mails. marco -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (Darwin) iEYEARECAAYFAkleWj8ACgkQ5A1A6ak/SoWoqACfVPL0A0kGakN93oI0eajQD6+K fu4AnRMCIXTVtjibqXey7N2SIYdmXt4t =eXZi -----END PGP SIGNATURE----- From kumarkm at tataelxsi.co.in Sat Jan 3 13:10:50 2009 From: kumarkm at tataelxsi.co.in (Kumar) Date: Sat, 3 Jan 2009 17:40:50 +0530 Subject: [gst-devel] G711 & G726 payload packetizer merging In-Reply-To: <000101c96cf2$5ef527a0$60033c0a@telxsi.com> Message-ID: <000601c96d9c$52720660$60033c0a@telxsi.com> Hi Wim, I am trying to stream G726 16,24,32,40kbps from gstreamer to VLC . 16 & 32 works fine but 24 & 40kbps is not proper(noisy). The pipeline that I am using is: Working: gst-launch-0.10 gstrtpbin name=rtpbin filesrc location=/home/ES_G726_32kbps.G726 ! audio/x-adpcm, rate=8000, channels=1, bitrate=32000, layout=g726 ! rtpg726pay ! rtpbin.send_rtp_sink_2 rtpbin.send_rtp_src_2 ! udpsink port=5002 host=10.60.3.96 name=vrtpsink rtpbin.send_rtcp_src_2 ! udpsink port=5002 host=10.60.3.96 sync=false name=vrtcpsink udpsrc port=5002 name=vrtpsrc ! rtpbin.recv_rtcp_sink_2 Noisy: 1) gst-launch-0.10 gstrtpbin name=rtpbin filesrc location=/home/ES_G726_24kbps.G726 ! audio/x-adpcm, rate=8000, channels=1, bitrate=24000, layout=g726 ! rtpg726pay ! rtpbin.send_rtp_sink_2 rtpbin.send_rtp_src_2 ! udpsink port=5002 host=10.60.3.96 name=vrtpsink rtpbin.send_rtcp_src_2 ! udpsink port=5002 host=10.60.3.96 sync=false name=vrtcpsink udpsrc port=5002 name=vrtpsrc ! rtpbin.recv_rtcp_sink_2 2) gst-launch-0.10 gstrtpbin name=rtpbin filesrc location=/home/ES_G726_40kbps.G726 ! audio/x-adpcm, rate=8000, channels=1, bitrate=40000, layout=g726 ! rtpg726pay ! rtpbin.send_rtp_sink_2 rtpbin.send_rtp_src_2 ! udpsink port=5002 host=10.60.3.96 name=vrtpsink rtpbin.send_rtcp_src_2 ! udpsink port=5002 host=10.60.3.96 sync=false name=vrtcpsink udpsrc port=5002 name=vrtpsrc ! rtpbin.recv_rtcp_sink_2 Whats is wrong in this pipeline? Thanks, Kumar KM -----Original Message----- From: Kumar [mailto:kumarkm at tataelxsi.co.in] Sent: Friday, January 02, 2009 9:24 PM To: 'Wim Taymans' Cc: 'Discussion of the development of GStreamer' Subject: Re: [gst-devel] G711 & G726 payload packetizer merging Ya. got it. Thanks very much Wim. :) T=Regards, Kumar KM -----Original Message----- From: Wim Taymans [mailto:wim.taymans at gmail.com] Sent: Friday, January 02, 2009 9:00 PM To: kumarkm at tataelxsi.co.in Cc: 'Discussion of the development of GStreamer' Subject: RE: [gst-devel] G711 & G726 payload packetizer merging On Fri, 2009-01-02 at 20:56 +0530, Kumar wrote: > Hi Wim, > > I could not find the source for the G711 RTP pay, but G726 payload > packitizer is present in the gst-plugins-good 0.10.10 package. > After installation of the package, gst-inspect will show only gstrtpg726 > pay. can you pls give me more info on this. rtppcma(de)pay/rtppcmu(de)pay are the G.711 a-law and mu-law (de)payloaders respectively. Wim > > Thanks , > > Regards, > Kumar KM > > -----Original Message----- > From: Wim Taymans [mailto:wim.taymans at gmail.com] > Sent: Friday, January 02, 2009 8:35 PM > To: kumarkm at tataelxsi.co.in; Discussion of the development of GStreamer > Subject: Re: [gst-devel] G711 & G726 payload packetizer merging > > > On Fri, 2009-01-02 at 12:38 +0530, Kumar wrote: > > Hi all, > > > > The G711 & G726 RTP payload packetizers are present in different > > gst-good-plugin packages. ( 0.10.2 & 0.10.10). > > Currently I'm merging G711 from Good 0.10.2 to Good 0.10.10. After making > > logical changes in Makefile.in & Makefile.am in gst/rtp directory of Good > > 0.10.10 plug-in package, > > I'm getting the following compilation error: > > > > libtool: link: `libgstrtp_la-gstrtpg711pay.lo' is not a valid libtool obje > > > > can anyone tell, why this error is coming? > > Is there any gst-good plug-in package which both G711 & G726 RTP payload > > packetizers are present? > > gst-plugins-good 0.10.10 has both elements. > > Wim > > > > > Thanks in advance.. > > > > Regards, > > Kumar KM > > > > > > The information contained in this electronic message and any attachments > to this message are intended for the exclusive use of the addressee(s) and > may contain proprietary, confidential or privileged information. If you are > not the intended recipient, you should not disseminate, distribute or copy > this e-mail. Please notify the sender immediately and destroy all copies of > this message and any attachments contained in it. > > > > -------------------------------------------------------------------------- > ---- > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. ---------------------------------------------------------------------------- -- _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From irfanshaikh at tataelxsi.co.in Sun Jan 4 14:18:52 2009 From: irfanshaikh at tataelxsi.co.in (Irfan Shaikh) Date: Sun, 4 Jan 2009 18:48:52 +0530 Subject: [gst-devel] G726 (24kbps and 32Kbps) Message-ID: <9D5E1752379A43408015F7FE98466115048C09D9@CHNEXVS01.VSNLXCHANGE.COM> Hi developers, I tried muxing G726(16kbps and 32kbps) in ASF container with ffmux_asf. It plays fine on locally on VLC player. But when I try the muxing G726(24kbps and 40kbs) in ASF container, o/p ASF file is generated. But The o/p ASF file plays locally on VLC player with noise. Can anyone please provide me some hint me to solve the problem ? Regards, This message (including any attachment) is confidential and may be legally privileged. Access to this message by anyone other than the intended recipient(s) listed above is unauthorized. If you are not the intended recipient you are hereby notified that any disclosure, copying, or distribution of the message, or any action taken or omission of action by you in reliance upon it, is prohibited and may be unlawful. Please immediately notify the sender by reply e-mail and permanently delete all copies of the message if you have received this message in error. -------------- next part -------------- An HTML attachment was scrubbed... URL: From andres.colubri at gmail.com Mon Jan 5 00:29:30 2009 From: andres.colubri at gmail.com (Andres Colubri) Date: Sun, 04 Jan 2009 15:29:30 -0800 Subject: [gst-devel] appsrc and appsink elements In-Reply-To: <494215F7.7050908@livecliq.com> References: <48903C57.2000904@st.com> <1217774661.23199.4.camel@metal> <4941D9C5.60909@gmail.com> <494215F7.7050908@livecliq.com> Message-ID: <4961465A.4010302@gmail.com> Hi, I've been struggling for a while with the appsrc element, and I wonder if anyone has any ideas about this. Basically, I'm trying to capture the contents of the screen into a video file. I'm not sure how to construct the pipeline to do it. I know that it has to start with an appsrc and end at a filesink, but I don't know how to initialize the appsrc (caps, max-bytes, etc) and what other intermediate elements are needed (encoders, demuxers). I looked at the examples that come with the source of appsrc, but these are for different scenarios (reading data from a file, streaming). Any help will be greatly appreciated. Thanks, Andres Jure Zitnik wrote: > Hi Andres, > The appsrc and appsink are not part of the binary gstreamer-plugins-bad > package on ubuntu 8.10 (reasoning is that they "still randomly change > API and ABI" :-) > But you can always build them from source: > apt-get source gstreamer0.10-plugins-bad > cd gst-plugins-bad0.10-0.10.5 (you might need to cange the version here) > ./configure > make > > After successful build, you can copy required shared libraries manually > (if you just need appsrc/appsink) as follows: > cp gst/app/.libs/libgstapp.so /usr/lib/gstreamer-0.10/ > cp gst-libs/gst/app/.libs/libgstapp-0.10.so.0.0.0 /usr/local/lib > You might need to create additional symbolic links in /usr/local/lib > (both linking to libgstapp-0.10.so.0.0.0) : libgstapp-0.10.so.0 and > libgstapp-0.10.so > > Now use gst-inspect to see if everything works fine... > > Jure > > Andres Colubri wrote: > >> Hello, >> >> I have tried to use appsrc, but for some reason it doesn't seem to be >> recognized (I'm using gst-plugins-bad 0.10.8 on ubuntu 0.8.10) by >> gstreamer. For example, when I run the appsrc_ex example, I get an >> assertion error right after >> app->src = gst_element_factory_make("appsrc", NULL); >> so seems that app->src is null >> >> Also, gst-inspect doesn't find neither appsrc nor appsink. >> >> Thanks >> Andres >> >> Wim Taymans wrote: >> >> >>> On Wed, 2008-07-30 at 12:03 +0200, David SIORPAES wrote: >>> >>> >>> >>>> Hello, >>>> >>>> I would like to exchange buffers which are allocated and managed by an >>>> application of mine to and from a GStreamer pipeline. >>>> >>>> By reading the documentation it looks like the appsrc/appsink are the >>>> solution for me, but I am also reading here >>>> http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/section-data-spoof.html >>>> that >>>> >>>> "New API is being developed at the moment to make data insertion and >>>> extraction less painful for applications. It can be found as GstAppSrc >>>> and GstAppSink in the gst-plugins-bad module. At the time of writing >>>> (October 2007), this API is not quite stable and ready yet, even though >>>> it may work fine for your purposes." >>>> >>>> So, may I ask whether it is safe to use such elements or not ? Or are >>>> there alternative solutions ? >>>> >>>> >>>> >>> The versionsof appsink/appsrc in the latest release of gst-plugins-bad >>> are full-featured and ready for consumption. See >>> http://webcvs.freedesktop.org/gstreamer/gst-plugins-bad/examples/app/ >>> for a whole bunch of examples of different use cases. >>> >>> Wim >>> >>> >>> >>>> Thanks, >>>> >>>> David S. >>>> >>>> >>>> >>>> >>>> ------------------------------------------------------------------------- >>>> This SF.Net email is sponsored by the Moblin Your Move Developer's challenge >>>> Build the coolest Linux based applications with Moblin SDK & win great prizes >>>> Grand prize is a trip for two to an Open Source event anywhere in the world >>>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> gstreamer-devel at lists.sourceforge.net >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>>> >>>> >>> ------------------------------------------------------------------------- >>> This SF.Net email is sponsored by the Moblin Your Move Developer's challenge >>> Build the coolest Linux based applications with Moblin SDK & win great prizes >>> Grand prize is a trip for two to an Open Source event anywhere in the world >>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >>> >>> >> ------------------------------------------------------------------------------ >> SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. >> The future of the web can't happen without you. Join us at MIX09 to help >> pave the way to the Next Web now. Learn more and register at >> http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > ------------------------------------------------------------------------------ > SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. > The future of the web can't happen without you. Join us at MIX09 to help > pave the way to the Next Web now. Learn more and register at > http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From gustavo.orrillo at gmail.com Mon Jan 5 00:43:33 2009 From: gustavo.orrillo at gmail.com (Gustavo Orrillo) Date: Sun, 4 Jan 2009 20:43:33 -0300 Subject: [gst-devel] appsrc and appsink elements In-Reply-To: <4961465A.4010302@gmail.com> References: <48903C57.2000904@st.com> <1217774661.23199.4.camel@metal> <4941D9C5.60909@gmail.com> <494215F7.7050908@livecliq.com> <4961465A.4010302@gmail.com> Message-ID: que estas tratando de hacer? 2009/1/4 Andres Colubri > Hi, > > I've been struggling for a while with the appsrc element, and I wonder > if anyone has any ideas about this. Basically, I'm trying to capture the > contents of the screen into a video file. I'm not sure how to construct > the pipeline to do it. I know that it has to start with an appsrc and > end at a filesink, but I don't know how to initialize the appsrc (caps, > max-bytes, etc) and what other intermediate elements are needed > (encoders, demuxers). I looked at the examples that come with the source > of appsrc, but these are for different scenarios (reading data from a > file, streaming). > > Any help will be greatly appreciated. > > Thanks, > Andres > > > Jure Zitnik wrote: > > Hi Andres, > > The appsrc and appsink are not part of the binary gstreamer-plugins-bad > > package on ubuntu 8.10 (reasoning is that they "still randomly change > > API and ABI" :-) > > But you can always build them from source: > > apt-get source gstreamer0.10-plugins-bad > > cd gst-plugins-bad0.10-0.10.5 (you might need to cange the version here) > > ./configure > > make > > > > After successful build, you can copy required shared libraries manually > > (if you just need appsrc/appsink) as follows: > > cp gst/app/.libs/libgstapp.so /usr/lib/gstreamer-0.10/ > > cp gst-libs/gst/app/.libs/libgstapp-0.10.so.0.0.0 /usr/local/lib > > You might need to create additional symbolic links in /usr/local/lib > > (both linking to libgstapp-0.10.so.0.0.0) : libgstapp-0.10.so.0 and > > libgstapp-0.10.so > > > > Now use gst-inspect to see if everything works fine... > > > > Jure > > > > Andres Colubri wrote: > > > >> Hello, > >> > >> I have tried to use appsrc, but for some reason it doesn't seem to be > >> recognized (I'm using gst-plugins-bad 0.10.8 on ubuntu 0.8.10) by > >> gstreamer. For example, when I run the appsrc_ex example, I get an > >> assertion error right after > >> app->src = gst_element_factory_make("appsrc", NULL); > >> so seems that app->src is null > >> > >> Also, gst-inspect doesn't find neither appsrc nor appsink. > >> > >> Thanks > >> Andres > >> > >> Wim Taymans wrote: > >> > >> > >>> On Wed, 2008-07-30 at 12:03 +0200, David SIORPAES wrote: > >>> > >>> > >>> > >>>> Hello, > >>>> > >>>> I would like to exchange buffers which are allocated and managed by an > >>>> application of mine to and from a GStreamer pipeline. > >>>> > >>>> By reading the documentation it looks like the appsrc/appsink are the > >>>> solution for me, but I am also reading here > >>>> > http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/section-data-spoof.html > >>>> that > >>>> > >>>> "New API is being developed at the moment to make data insertion and > >>>> extraction less painful for applications. It can be found as GstAppSrc > >>>> and GstAppSink in the gst-plugins-bad module. At the time of writing > >>>> (October 2007), this API is not quite stable and ready yet, even > though > >>>> it may work fine for your purposes." > >>>> > >>>> So, may I ask whether it is safe to use such elements or not ? Or are > >>>> there alternative solutions ? > >>>> > >>>> > >>>> > >>> The versionsof appsink/appsrc in the latest release of gst-plugins-bad > >>> are full-featured and ready for consumption. See > >>> http://webcvs.freedesktop.org/gstreamer/gst-plugins-bad/examples/app/ > >>> for a whole bunch of examples of different use cases. > >>> > >>> Wim > >>> > >>> > >>> > >>>> Thanks, > >>>> > >>>> David S. > >>>> > >>>> > >>>> > >>>> > >>>> > ------------------------------------------------------------------------- > >>>> This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > >>>> Build the coolest Linux based applications with Moblin SDK & win great > prizes > >>>> Grand prize is a trip for two to an Open Source event anywhere in the > world > >>>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ > >>>> _______________________________________________ > >>>> gstreamer-devel mailing list > >>>> gstreamer-devel at lists.sourceforge.net > >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > >>>> > >>>> > >>>> > >>> > ------------------------------------------------------------------------- > >>> This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > >>> Build the coolest Linux based applications with Moblin SDK & win great > prizes > >>> Grand prize is a trip for two to an Open Source event anywhere in the > world > >>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ > >>> _______________________________________________ > >>> gstreamer-devel mailing list > >>> gstreamer-devel at lists.sourceforge.net > >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > >>> > >>> > >>> > >>> > >> > ------------------------------------------------------------------------------ > >> SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, > Nevada. > >> The future of the web can't happen without you. Join us at MIX09 to > help > >> pave the way to the Next Web now. Learn more and register at > >> > http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ > >> _______________________________________________ > >> gstreamer-devel mailing list > >> gstreamer-devel at lists.sourceforge.net > >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > >> > >> > > > > > > > ------------------------------------------------------------------------------ > > SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, > Nevada. > > The future of the web can't happen without you. Join us at MIX09 to help > > pave the way to the Next Web now. Learn more and register at > > > http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From yangsb05 at gmail.com Mon Jan 5 05:02:58 2009 From: yangsb05 at gmail.com (yang shaobo) Date: Mon, 5 Jan 2009 12:02:58 +0800 Subject: [gst-devel] How to install v4l2 plugin Message-ID: <4feed6860901042002u75d97e47ud48e6478b8944998@mail.gmail.com> I find the v4l2src plugin in gst-plugin-good. I want to try it on my board. When I execute ./configure --enable_gst_v4l2 , I found the following messages: configure: *** checking feature: Video 4 Linux 2 *** configure: *** for plug-ins: v4l2src *** configure: WARNING: *** v4l2src pre-configured not to be built configure: *** These plugins will not be built: v4l2src checking for X... libraries , headers I add the --enable_gst_v4l2 option when executing configure ! Can anyone give me some advice ? From vaisakh.n at nestgroup.net Mon Jan 5 05:16:58 2009 From: vaisakh.n at nestgroup.net (vaisakh.n) Date: Mon, 05 Jan 2009 09:46:58 +0530 Subject: [gst-devel] Hi.. Message-ID: <496189BA.8040900@nestgroup.net> I tried to run helloworld program from gstreamer application developers manual link: http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/chapter-helloworld.html but its it hangs after showning:: Now playing :"file name" thats it get hanged in function gst_element_set_state (pipeline, GST_STATE_PLAYING); what to do .. please help me.. advance thanks.. ***** Confidentiality Statement/Disclaimer ***** This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. From d.valeswararao at gmail.com Mon Jan 5 05:45:32 2009 From: d.valeswararao at gmail.com (Valeswararao Dhulipudi) Date: Mon, 05 Jan 2009 04:45:32 +0000 Subject: [gst-devel] New Birthday Calendar Message-ID: Hello I am creating a birthday calendar for myself. Can you please click on the link below and enter your birthday for me? http://www.birthdayalarm.com/bd2/83940861a169585367b1443039839c505612347d1386 Valeswararao From vaisakh.n at nestgroup.net Mon Jan 5 07:46:32 2009 From: vaisakh.n at nestgroup.net (vaisakh.n) Date: Mon, 05 Jan 2009 12:16:32 +0530 Subject: [gst-devel] Hi Message-ID: <4961ACC8.6040008@nestgroup.net> help me to use gst-editor-0.8. I created xml file like this ./gst-launch-0.10 filesrc location=KDE_Logout_1.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! osssink -o test3.xml and started the gst-editor like ./gst-editor test3.xml but it shows error like this /**********************************************/ ** Message: palette visible: TRUE ** Message: making new palette (gst-editor:5188): GStreamer-CRITICAL **: file gstxml.c: line 415 (gst_xml_make_element): assertion `element != NULL' failed (gst-editor:5188): GStreamer-CRITICAL **: file gstobject.c: line 223 (gst_object_ref): assertion `GST_IS_OBJECT (object)' failed (gst-editor:5188): GStreamer-CRITICAL **: file gstobject.c: line 563 (gst_object_unparent): assertion `object != NULL' failed (gst-editor:5188): GStreamer-CRITICAL **: file gstbin.c: line 520 (gst_bin_add): assertion `GST_IS_ELEMENT (element)' failed ** Message: linking GUI for vorbisdec0:src and audioconvert0:sink ** Message: linking GUI for filesrc0:src and oggdemux0:sink /************************************************/ and in gst-editor gui no proper links among the elements help me... thanks.. ***** Confidentiality Statement/Disclaimer ***** This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. From vaisakh.n at nestgroup.net Mon Jan 5 09:52:01 2009 From: vaisakh.n at nestgroup.net (vaisakh.n) Date: Mon, 05 Jan 2009 14:22:01 +0530 Subject: [gst-devel] hi Message-ID: <4961CA31.8000806@nestgroup.net> gst-editor 0.8 help... how can i add mad plug in to gst-editor,.. that element not showing in gst-editor eventhough i installed the mad plug in and heared mp3 using it ... (via gst-launch -0.10 tool).. help me.. advance thanks.. ***** Confidentiality Statement/Disclaimer ***** This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. From bilboed at gmail.com Mon Jan 5 10:27:02 2009 From: bilboed at gmail.com (Edward Hervey) Date: Mon, 05 Jan 2009 10:27:02 +0100 Subject: [gst-devel] How to install v4l2 plugin In-Reply-To: <4feed6860901042002u75d97e47ud48e6478b8944998@mail.gmail.com> References: <4feed6860901042002u75d97e47ud48e6478b8944998@mail.gmail.com> Message-ID: <1231147622.14671.2.camel@joder> Hi, You need to build -good with --enable-experimental. Edward On Mon, 2009-01-05 at 12:02 +0800, yang shaobo wrote: > I find the v4l2src plugin in gst-plugin-good. > I want to try it on my board. > When I execute ./configure --enable_gst_v4l2 , I found the following messages: > > configure: *** checking feature: Video 4 Linux 2 *** > configure: *** for plug-ins: v4l2src *** > configure: WARNING: *** v4l2src pre-configured not to be built > configure: *** These plugins will not be built: v4l2src > checking for X... libraries , headers > > I add the --enable_gst_v4l2 option when executing configure ! > Can anyone give me some advice ? > > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From bilboed at gmail.com Mon Jan 5 10:27:57 2009 From: bilboed at gmail.com (Edward Hervey) Date: Mon, 05 Jan 2009 10:27:57 +0100 Subject: [gst-devel] New Birthday Calendar In-Reply-To: References: Message-ID: <1231147677.14671.3.camel@joder> err... What does this have to do with gstreamer development ? Who are you ? Edward On Mon, 2009-01-05 at 04:45 +0000, Valeswararao Dhulipudi wrote: > Hello > > I am creating a birthday calendar for myself. Can you please click on the link below and enter your birthday for me? > > http://www.birthdayalarm.com/bd2/83940861a169585367b1443039839c505612347d1386 > > Valeswararao > > > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From d.valeswararao at gmail.com Mon Jan 5 11:12:29 2009 From: d.valeswararao at gmail.com (valeswara rao) Date: Mon, 5 Jan 2009 15:42:29 +0530 Subject: [gst-devel] New Birthday Calendar Message-ID: hi bilboed that mial is spam by automatically send to all my mail contacts....pelase delete it. thanks®ards Valeswararao .D -------------- next part -------------- An HTML attachment was scrubbed... URL: From raffel.mk at gmail.com Fri Jan 2 12:05:38 2009 From: raffel.mk at gmail.com (Lin Raffel) Date: Fri, 2 Jan 2009 19:05:38 +0800 Subject: [gst-devel] otem-gstreamer-video-thumbnailer sometimes fails to generate the thumbnail of ASF file Message-ID: I found totem-gstreamer-video-thumbnailer sometimes fails to generate the thumbnail of ASF file, the output message is. Reason: Took too much time to process. Is this the problem with the ASF file itself ? -------------- next part -------------- An HTML attachment was scrubbed... URL: From sameer at nextbitcpu.com Mon Jan 5 11:24:06 2009 From: sameer at nextbitcpu.com (Sameer Naik) Date: Mon, 5 Jan 2009 15:54:06 +0530 Subject: [gst-devel] Hi In-Reply-To: <4961ACC8.6040008@nestgroup.net> References: <4961ACC8.6040008@nestgroup.net> Message-ID: <200901051554.06534.sameer@nextbitcpu.com> First of gst-editor-0.8 is meant for use with gstreamer-0.8 and is not ported for gstreamer-0.10. Secondly, please choose an appropriate subject for your mails. Regards ~Sameer On Monday 05 January 2009 12:16:32 pm vaisakh.n wrote: > help me to use gst-editor-0.8. > I created xml file like this > > ./gst-launch-0.10 filesrc location=KDE_Logout_1.ogg ! oggdemux ! > vorbisdec ! audioconvert ! audioresample ! osssink -o test3.xml > > and > started the gst-editor like > ./gst-editor test3.xml > > > but it shows error > like this > > /**********************************************/ > > > > ** Message: palette visible: TRUE > ** Message: making new palette > > (gst-editor:5188): GStreamer-CRITICAL **: file gstxml.c: line 415 > (gst_xml_make_element): assertion `element != NULL' failed > > (gst-editor:5188): GStreamer-CRITICAL **: file gstobject.c: line 223 > (gst_object_ref): assertion `GST_IS_OBJECT (object)' failed > > (gst-editor:5188): GStreamer-CRITICAL **: file gstobject.c: line 563 > (gst_object_unparent): assertion `object != NULL' failed > > (gst-editor:5188): GStreamer-CRITICAL **: file gstbin.c: line 520 > (gst_bin_add): assertion `GST_IS_ELEMENT (element)' failed > ** Message: linking GUI for vorbisdec0:src and audioconvert0:sink > ** Message: linking GUI for filesrc0:src and oggdemux0:sink > /************************************************/ > > and in gst-editor gui no proper links among the elements > > > help me... > thanks.. > > ***** Confidentiality Statement/Disclaimer ***** > > This message and any attachments is intended for the sole use of the > intended recipient. It may contain confidential information. Any > unauthorized use, dissemination or modification is strictly prohibited. If > you are not the intended recipient, please notify the sender immediately > then delete it from all your systems, and do not copy, use or print. > Internet communications are not secure and it is the responsibility of the > recipient to make sure that it is virus/malicious code exempt. The > company/sender cannot be responsible for any unauthorized alterations or > modifications made to the contents. If you require any form of confirmation > of the contents, please contact the company/sender. The company/sender is > not liable for any errors or omissions in the content of this message. > > --------------------------------------------------------------------------- >--- _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From Nagamani.Veerappa at csr.com Mon Jan 5 11:41:46 2009 From: Nagamani.Veerappa at csr.com (Nagamani Veerappa) Date: Mon, 5 Jan 2009 16:11:46 +0530 Subject: [gst-devel] Is there any UPnP Media sever that can share real audio on RTSP? Message-ID: <610EAA62A0FD314D9761697EEC820CBE64A076@banasiexb01.ASIA.ROOT.PRI> Hi, Is there any Gstreamer application/Plugin or anything else, that can work like UPnP Media server sharing real audio on RTSP? Please let me know. Thanks Nagamani -------------- next part -------------- An HTML attachment was scrubbed... URL: From wierdo_b at hotmail.com Mon Jan 5 14:07:24 2009 From: wierdo_b at hotmail.com (Wierd O) Date: Mon, 5 Jan 2009 14:07:24 +0100 Subject: [gst-devel] Error in parsing:corrupted stream Message-ID: Hi all, I am using gstreamer to play video and come up with an error on the decoder. Here is the error ... ################################################################ root at mx21ads:~$ gst-launch filesrc -v location=/video/test_5.mp4 ! mfw_mp4demuxer name=demux demux. ! mfw_vpudecoder codec-type=std_avc ! mfw_v4lsink demux. & root at mx21ads:~$ Setting pipeline to PAUSED ... ERROR: Pipeline doesn't want to pause. ERROR: from element /pipeline0/demux: Error in parsing:corrupted stream Additional debug info: debug none Setting pipeline to NULL ... FREEING pipeline ... ################################################################ What could be the problem ? Thanks in advance , W. _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From bisht.sudarshan at gmail.com Mon Jan 5 14:24:14 2009 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Mon, 5 Jan 2009 18:54:14 +0530 Subject: [gst-devel] Error in parsing:corrupted stream In-Reply-To: References: Message-ID: <785339900901050524m28dab71r9e545a5fe4cb3c39@mail.gmail.com> Hi , Try playing with qtdemux plugin. On Mon, Jan 5, 2009 at 6:37 PM, Wierd O wrote: > > Hi all, > > I am using gstreamer to play video and come up with an error on the > decoder. Here is the error ... > > ################################################################ > > root at mx21ads:~$ gst-launch filesrc -v location=/video/test_5.mp4 ! > mfw_mp4demuxer name=demux demux. ! mfw_vpudecoder codec-type=std_avc ! > mfw_v4lsink demux. & > root at mx21ads:~$ Setting pipeline to PAUSED ... > ERROR: Pipeline doesn't want to pause. > *ERROR: from element /pipeline0/demux: Error in parsing:corrupted stream* > Additional debug info: > debug none > Setting pipeline to NULL ... > FREEING pipeline ... > > > ################################################################ > > What could be the problem ? > > Thanks in advance , > > W. > > > > > ------------------------------ > Express yourself instantly with MSN Messenger! MSN Messenger > > > ------------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From thiagossantos at gmail.com Mon Jan 5 14:33:58 2009 From: thiagossantos at gmail.com (thiagoss) Date: Mon, 5 Jan 2009 11:33:58 -0200 Subject: [gst-devel] gstbin test... In-Reply-To: <956359.4169.qm@web70807.mail.kr3.yahoo.com> References: <956359.4169.qm@web70807.mail.kr3.yahoo.com> Message-ID: On Mon, Dec 29, 2008 at 2:51 PM, Yool-Je Cho wrote: > Hello... > > I tested gstbin.. > > gst-launch-0.10 audiotestsrc ! queue ! audioconvert ! { vorbisenc ! oggmux > } ! filesink location=audio.ogg > is successed... > > but...below is > Error: Internal data flow error. (gstbasesrc.c(2234): gst_base_src_loop (): > /GstPipeline:pipeline0/GstAudioTestSrc:audiotestsrc0: > streaming task paused, reason not-linked (-1)) > > > > static GstElement *make_encoder (void) > { > GstElement *result; > GstElement *codec; > GstElement *mux; > GstPad *pad; > GstCaps *caps; > > /* create result bin */ > result = gst_bin_new(NULL); > g_assert(result); > > /* create elements */ > codec = gst_element_factory_make("vorbisenc", NULL); > g_assert(codec); > > mux = gst_element_factory_make("oggmux", NULL); > g_assert(mux); > > /* add elements to result bin */ > gst_bin_add(GST_BIN(result), codec); > gst_bin_add(GST_BIN(result), mux); > > /* link elements */ > gst_element_link_pads(codec, "src", mux, "sink"); > You should have tested the result of this call, it is failing because 'mux' has no pad named 'sink'. oggmux uses 'request pads', read the docs about it and you'll see how to fix this. > /* ghost src and sink pads */ > pad = gst_element_get_static_pad(codec, "sink"); > gst_element_add_pad(result, gst_ghost_pad_new("sink", pad)); > gst_object_unref(pad); > > pad = gst_element_get_static_pad(mux, "src"); > gst_element_add_pad(result, gst_ghost_pad_new("src", pad)); > gst_object_unref(pad); > > return result; > } > > > static GstElement *make_pipeline(void) > { > GstElement *result; > GstElement *audiosrc; > GstElement *queue; > GstElement *aconv; > GstElement *sink; > GstElement *encoder; > > result = gst_pipeline_new(NULL); > g_assert(result); > > /* create various elements */ > audiosrc = gst_element_factory_make("audiotestsrc", NULL); > g_assert(audiosrc); > > queue = gst_element_factory_make("queue", NULL); > g_assert(queue); > > aconv = gst_element_factory_make("audioconvert", NULL); > g_assert(aconv); > > > sink = gst_element_factory_make("filesink", NULL); > g_object_set(sink, "location", "audio.ogg", NULL); > g_assert(sink); > > /* add elements */ > gst_bin_add(GST_BIN(result), audiosrc); > gst_bin_add(GST_BIN(result), queue); > gst_bin_add(GST_BIN(result), aconv); > gst_bin_add(GST_BIN(result), sink); > > encoder = make_encoder(); > > gst_bin_add(GST_BIN(result), encoder); > > /* link elements */ > gst_element_link_pads(audiosrc, "src", queue, "sink"); > gst_element_link_pads(queue, "src", aconv, "sink"); > gst_element_link_pads(aconv, "src", encoder, "sink"); > gst_element_link_pads(encoder, "src", sink, "sink"); > > gst_element_set_state(result, GST_STATE_READY); > > return result; > } > > > Where is incorrect? > > Thanks.. > > > ------------------------------ > > *180? ??? ??! ??* > ??! ??? ??! ????? > ???? ???? ??? ?? ? ????. > > > ------------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Thiago Sousa Santos Embedded Systems and Pervasive Computing Lab (Embedded) Center of Electrical Engineering and Informatics (CEEI) Federal University of Campina Grande (UFCG) -------------- next part -------------- An HTML attachment was scrubbed... URL: From wierdo_b at hotmail.com Mon Jan 5 14:47:08 2009 From: wierdo_b at hotmail.com (Wierd O) Date: Mon, 5 Jan 2009 14:47:08 +0100 Subject: [gst-devel] Error in parsing:corrupted stream In-Reply-To: <785339900901050524m28dab71r9e545a5fe4cb3c39@mail.gmail.com> References: <785339900901050524m28dab71r9e545a5fe4cb3c39@mail.gmail.com> Message-ID: Sudarshan: I have tried the Quick time demuxer too and it couldnt find the demux... ################################################################ root at mx21ads:~$ gst-launch filesrc -v location=/video/test_5.mp4 ! qtdemux name=demux demux. ! mfw_vpudecoder codec-type=std_avc ! mfw_v4lsink disp-width=800 disp-height=600 demux. & root at mx21ads:~$ Setting pipeline to PAUSED ... Pipeline is PREROLLING ... ERROR: from element /pipeline0/demux: GStreamer encountered a general stream error. Additional debug info: qtdemux.c(1617): gst_qtdemux_loop (): /pipeline0/demux: no known streams found ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... FREEING pipeline ... ################################################################## Date: Mon, 5 Jan 2009 18:54:14 +0530 From: bisht.sudarshan at gmail.com To: gstreamer-devel at lists.sourceforge.net Subject: Re: [gst-devel] Error in parsing:corrupted stream Hi , Try playing with qtdemux plugin. On Mon, Jan 5, 2009 at 6:37 PM, Wierd O wrote: Hi all, I am using gstreamer to play video and come up with an error on the decoder. Here is the error ... ################################################################## root at mx21ads:~$ gst-launch filesrc -v location=/video/test_5.mp4 ! mfw_mp4demuxer name=demux demux. ! mfw_vpudecoder codec-type=std_avc ! mfw_v4lsink demux. & root at mx21ads:~$ Setting pipeline to PAUSED ... ERROR: Pipeline doesn't want to pause. ERROR: from element /pipeline0/demux: Error in parsing:corrupted stream Additional debug info: debug none Setting pipeline to NULL ... FREEING pipeline ... ################################################################ What could be the problem ? Thanks in advance , W. Express yourself instantly with MSN Messenger! MSN Messenger ------------------------------------------------------------------------------ _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -- Regards, Sudarshan Bisht _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From jack.foster11 at gmail.com Mon Jan 5 15:55:03 2009 From: jack.foster11 at gmail.com (Jack Foster) Date: Mon, 5 Jan 2009 09:55:03 -0500 Subject: [gst-devel] RTP C Examples In-Reply-To: <1230909759.31475.58.camel@metal> References: <1230909759.31475.58.camel@metal> Message-ID: I think the picture is worth a thousand words. That makes it all come clear. Could that be added to the doc? Jack On Fri, Jan 2, 2009 at 10:22 AM, Wim Taymans wrote: > On Wed, 2008-12-31 at 11:03 -0500, Jack Foster wrote: > > Hi gang! > > > > I see lots of examples that have CLI to gStreamer but surely people > > are building applications with C. It seems like an awesome framework > > but I'm just trying to get my head around it. I would like to see > > more C examples. > > > > In any case, I'm trying to a simple audio rtp sender/receiver working > > but am having a hard time translating the example from the gstrtpbin > > documentation to C. I'll then post a working copy on the mailing list > > for others running into the same issue. > > > > I really have no idea if the errors are related to the mad plugin or > > the pads not linked together properly. > > > > Attached is the error and the program. Any help would be greatly > > appreciated. > > I added some C examples in CVS here: > > > http://webcvs.freedesktop.org/gstreamer/gst-plugins-good/tests/examples/rtp/ > > Look at client-PCMA.c and server-alsasrc-PCMA.c > > Wim > > > > > Thanks! > > Jack > > > ------------------------------------------------------------------------------ > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From msmith at xiph.org Mon Jan 5 18:52:15 2009 From: msmith at xiph.org (Michael Smith) Date: Mon, 5 Jan 2009 09:52:15 -0800 Subject: [gst-devel] hi In-Reply-To: <4961CA31.8000806@nestgroup.net> References: <4961CA31.8000806@nestgroup.net> Message-ID: <3c1737210901050952s4ef9a972j8e0b0ff20836e4e3@mail.gmail.com> gst-editor-0.8 is not compatible with gstreamer 0.10. There is no fully-functional gst-editor for gstreamer 0.10, but in practice this hasn't been a significant issue. Mike On Mon, Jan 5, 2009 at 12:52 AM, vaisakh.n wrote: > gst-editor 0.8 help... > how can i add mad plug in to gst-editor,.. > that element not showing in gst-editor > eventhough i installed the mad plug in and heared mp3 using it ... > (via gst-launch -0.10 tool).. > help me.. > advance thanks.. > > ***** Confidentiality Statement/Disclaimer ***** > > This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. > The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. > > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From msmith at xiph.org Mon Jan 5 18:55:43 2009 From: msmith at xiph.org (Michael Smith) Date: Mon, 5 Jan 2009 09:55:43 -0800 Subject: [gst-devel] Error in parsing:corrupted stream In-Reply-To: References: Message-ID: <3c1737210901050955i765d6ac5la0c9b6a5db4dbcd9@mail.gmail.com> On Mon, Jan 5, 2009 at 5:07 AM, Wierd O wrote: > > Hi all, > > I am using gstreamer to play video and come up with an error on the decoder. > Here is the error ... > > ################################################################ > > root at mx21ads:~$ gst-launch filesrc -v location=/video/test_5.mp4 ! > mfw_mp4demuxer name=demux demux. ! mfw_vpudecoder codec-type=std_avc ! > mfw_v4lsink demux. & > root at mx21ads:~$ Setting pipeline to PAUSED ... > ERROR: Pipeline doesn't want to pause. > ERROR: from element /pipeline0/demux: Error in parsing:corrupted stream > Additional debug info: > debug none > Setting pipeline to NULL ... > FREEING pipeline ... We obviously can't support these third-party closed-source plugins. An equivalent using open-source plugins might be something like this: gst-launch-0.10 filesrc location=test_5.mp4 ! decodebin ! ffmpegcolorspace ! videoscale ! xvimagesink (internally, decodebin will be using qtdemux and an appropriate autoplugged decoder). If that doesn't work, then either you don't have appropriate plugins installed, or there's something wrong with your file (or you've encountered a bug - but that's relatively unlikely). Mike From julien.isorce at gmail.com Mon Jan 5 23:37:38 2009 From: julien.isorce at gmail.com (Julien Isorce) Date: Mon, 5 Jan 2009 23:37:38 +0100 Subject: [gst-devel] physical inputs, audio source Message-ID: <180a127d0901051437s3138bf47p8abee052f944722b@mail.gmail.com> Hi, http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gsttuner.html#gst-tuner-list-channelsis used to list (and select) the physical (video) inputs of a tuner card. What about selecting physical inputs of an audio source device ? How alsasrc does it ? (GstMixerTrack represents a physical input ?) (gst_mixer_set_record () select a physical input ?) Thx Sincerely Julien -------------- next part -------------- An HTML attachment was scrubbed... URL: From julien.isorce at gmail.com Mon Jan 5 23:44:05 2009 From: julien.isorce at gmail.com (Julien Isorce) Date: Mon, 5 Jan 2009 23:44:05 +0100 Subject: [gst-devel] iphone/ipodtouch Message-ID: <180a127d0901051444p3ef15e23x7547169ba62d9b66@mail.gmail.com> Hi Does someone tried to compile then run gstreamer on iphone/ipodtouch ? (I have no Mac to try but I have an ipod touch, I hope to find a way to develop/test gst-gl on it but first I am wondering if this is interesting ?) Sincerely Julien -------------- next part -------------- An HTML attachment was scrubbed... URL: From johngro at google.com Mon Jan 5 23:49:36 2009 From: johngro at google.com (John Grossman) Date: Mon, 5 Jan 2009 14:49:36 -0800 Subject: [gst-devel] iphone/ipodtouch In-Reply-To: <180a127d0901051444p3ef15e23x7547169ba62d9b66@mail.gmail.com> References: <180a127d0901051444p3ef15e23x7547169ba62d9b66@mail.gmail.com> Message-ID: Currently I believe that the public iPhone and iPodTouch development environments are ObjectiveC only. I imagine that this will cause trouble when attempting to port the gstreamer framework to the platform. peace -john On Mon, Jan 5, 2009 at 2:44 PM, Julien Isorce wrote: > Hi > > Does someone tried to compile then run gstreamer on iphone/ipodtouch ? > > (I have no Mac to try but I have an ipod touch, I hope to find a way to > develop/test gst-gl on it but first I am wondering if this is interesting ?) > > Sincerely > > Julien > > > ------------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From thx1138 at earthlink.net Tue Jan 6 01:48:15 2009 From: thx1138 at earthlink.net (thx1138) Date: Mon, 05 Jan 2009 16:48:15 -0800 Subject: [gst-devel] VST Audio Plug in approach using Gstreamer and Linux OS with ARM Cortex A8 processors Message-ID: Hello, We are presently using VST as a plug-in for audio applications and the question came up today is it possible for VST plug-ins to work with G-Streamer? Our VST's are migrating from PC/Mac Intel host and desire to use VST approach with Gstreamer for ARM Cortex A8 processor. We have Linux Gstreamer as our Linux interface and we are wondering what has been done in this area already to look at the API interfaces etc. Regards, Terry From bilboed at gmail.com Tue Jan 6 10:23:28 2009 From: bilboed at gmail.com (Edward Hervey) Date: Tue, 06 Jan 2009 10:23:28 +0100 Subject: [gst-devel] iphone/ipodtouch In-Reply-To: <180a127d0901051444p3ef15e23x7547169ba62d9b66@mail.gmail.com> References: <180a127d0901051444p3ef15e23x7547169ba62d9b66@mail.gmail.com> Message-ID: <1231233809.9864.2.camel@joder> Hi, I managed to compile gstreamer+dependencies and quite a few plugins for unlocked iphone some time back (before the public *sdk* was available). I didn't try video output, but wrote some audio source/sinks using the ultra-basic AudioQueue API as subclasses of GstBaseAudio[Sink|Src]. No idea how far you could get with a standard/locked iphone/ipodtouch though. Edward On Mon, 2009-01-05 at 23:44 +0100, Julien Isorce wrote: > Hi > > Does someone tried to compile then run gstreamer on iphone/ipodtouch ? > > (I have no Mac to try but I have an ipod touch, I hope to find a way > to develop/test gst-gl on it but first I am wondering if this is > interesting ?) > > Sincerely > > Julien > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From liangzhihong1984 at 126.com Tue Jan 6 10:48:59 2009 From: liangzhihong1984 at 126.com (liangzhihong1984) Date: Tue, 6 Jan 2009 17:48:59 +0800 (CST) Subject: [gst-devel] About buffer size problem in udpsrc element Message-ID: <12351133.745051231235339064.JavaMail.coremail@bj126app56.126.com> Hi, everyone. I was using udpsrc to receive mpeg2ts stream from other machines, But it's not so good when displaying. And I got some information from the gstudpsrc.c file that "the default size is typically 50k and can be increased to maximally 100k", so I think the buffer size is not enough for my application, now I want to expand the buffer size to a certain value, what should I do? Thank you all. Best Regards, Leon Liang -------------- next part -------------- An HTML attachment was scrubbed... URL: From niej0001 at gmail.com Tue Jan 6 11:13:59 2009 From: niej0001 at gmail.com (Nie Jun) Date: Tue, 6 Jan 2009 18:13:59 +0800 Subject: [gst-devel] About buffer size problem in udpsrc element In-Reply-To: <12351133.745051231235339064.JavaMail.coremail@bj126app56.126.com> References: <12351133.745051231235339064.JavaMail.coremail@bj126app56.126.com> Message-ID: <7c34ac520901060213mbaa5f37u596be18dc510b71b@mail.gmail.com> you can try property: blocksize and buffer-size to enable udpsrc emit larger buffer. you can gst-inspect udpsrc or check source code for their detail information. > > ------------------------------ > ???????????????????5000???? > > ------------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liangzhihong1984 at 126.com Tue Jan 6 11:52:04 2009 From: liangzhihong1984 at 126.com (liangzhihong1984) Date: Tue, 6 Jan 2009 18:52:04 +0800 (CST) Subject: [gst-devel] About buffer size problem in udpsrc element Message-ID: <18410693.762651231239124234.JavaMail.coremail@bj126app56.126.com> Thank you for your reply. Then what's the difference or relationship between the two properties blocksize and buffer-size? -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilboed at gmail.com Tue Jan 6 12:06:11 2009 From: bilboed at gmail.com (Edward Hervey) Date: Tue, 06 Jan 2009 12:06:11 +0100 Subject: [gst-devel] Conversion to GIT (no, really, I mean it this time) Message-ID: <1231239971.9864.24.camel@joder> Hi all, No, this is no longer another boring mail from myself regarding test repositories and all that crap... ... WE ARE REALLY SWITCHING TO GIT THIS MONTH ! The current switch date is set to just after the upcoming releases of core/base/python/bad, which should be on Monday 19th of January. What will happen at that point: * All of gstreamer cvs repositories will become read-only * All the gstreamer modules will be converted to git using the scripts used to do the previous git test conversions * The git repositories will be pushed as read-only to the main freedesktop.org servers. * After some final integrity checks (hopefully less than a few hours) the repositories will be made read/write * Go party * Make GStreamer even more awesome A big thanks to everyone involved for their patience regarding this conversion, it's been a hectic road since the decision back in Istanbul, but I think it was worth the ride. Edward From wim.taymans at gmail.com Tue Jan 6 12:10:05 2009 From: wim.taymans at gmail.com (Wim Taymans) Date: Tue, 6 Jan 2009 12:10:05 +0100 Subject: [gst-devel] About buffer size problem in udpsrc element In-Reply-To: <18410693.762651231239124234.JavaMail.coremail@bj126app56.126.com> References: <18410693.762651231239124234.JavaMail.coremail@bj126app56.126.com> Message-ID: blocksize is a default property for all sources, it's unused in udpsrc. buffer-size is the size of the kernel receive buffer, it allows the kernel to buffer more packets for when the app is not processing packets fast enough. 2009/1/6 liangzhihong1984 : > > Thank you for your reply. > Then what's the difference or relationship between the two properties > blocksize and buffer-size? > > ________________________________ > ???????????????????5000???? > ------------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From rahul.verma at sasken.com Tue Jan 6 11:40:36 2009 From: rahul.verma at sasken.com (Rahul Verma) Date: Tue, 6 Jan 2009 16:10:36 +0530 Subject: [gst-devel] 3gp demux plugin working on PC but not on target Message-ID: <89635EF2797FED4B9EF9769F80CEABC81AF3B1@EXGZ01.sasken.com> Hi, I am developing a 3gp Demux plugin. My pipeline is filesrc ! 3gpreader(custom) ! mpeg4decoder(custom) ! xvimagesink. This works properly on the PC. But when I try to run the same pipeline on target filesrc ! 3gpreader(custom) ! mpeg4decoder(custom) ! v4l2videosink, it does not work. I get the following error : gst_bus_new: created new bus Caught SIGSEGV accessing address (nil) Spinning. Please run 'gdb gst-launch 129' to continue debugging, Ctrl-C to quit, or Ctrl-\ to dump core. After checking the source files it is while attaching the bus to the pipeline. However pipeline filesrc ! ffdemux_mov_mp4_m4a_3gp_3g2_mj2 ! mpeg4decoder(custom) ! v4l2videosink works fine with GST_DEBUG=0. In this pipeline also if I set GST_DEBUG=3/4/5 i get the same SIGSEGV fault at the same point. I am stuck here and need some pointers. Regards, Rahul SASKEN BUSINESS DISCLAIMER ------------------------- This message may contain confidential, proprietary or legally privileged information. 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URL: From dstaples at friendmts.co.uk Tue Jan 6 13:20:34 2009 From: dstaples at friendmts.co.uk (Darren Staples) Date: Tue, 6 Jan 2009 12:20:34 -0000 Subject: [gst-devel] Message-ID: I?ve only recently come across Gstreamer, so I?m hoping you guys might be able to give me some pointers as to the best way to achieve the following. I need an application which will take video input (in any of the common i.e. AVI, FLV, MPG, MOV etc. formats) from a local file, and decode the frames one by one. I want the frames to be 8-bit luminance only i.e. greyscale, and I need to be able to access the raw pixel data for each frame. Processing of the luminance frame data may be slower than the OEnatural? frame rate of the video, so OEplayback? will almost certainly be slower than real-time. How do I control the pipeline so I process a single frame before decoding the next ? I don?t care about any audio track that might or might not be present. Any pointers or tips you can offer would be greatly appreciated :) Thanks in advance, Darren -------------- next part -------------- An HTML attachment was scrubbed... URL: From wim.taymans at gmail.com Tue Jan 6 13:42:53 2009 From: wim.taymans at gmail.com (Wim Taymans) Date: Tue, 6 Jan 2009 13:42:53 +0100 Subject: [gst-devel] In-Reply-To: References: Message-ID: The best way is to create a pipeline with uridecodebin as the generic decoder element and then link appsink as the sink element and pull buffers from it one by one. You can configure appsink to only accept grey video and use ffmpegcolorspace to convert to that format Wim On Tue, Jan 6, 2009 at 1:20 PM, Darren Staples wrote: > I've only recently come across Gstreamer, so I'm hoping you guys might be > able to give me some pointers as to the best way to achieve the following. > > I need an application which will take video input (in any of the common i.e. > AVI, FLV, MPG, MOV etc. formats) from a local file, and decode the frames > one by one. I want the frames to be 8-bit luminance only i.e. greyscale, and > I need to be able to access the raw pixel data for each frame. > > Processing of the luminance frame data may be slower than the 'natural' > frame rate of the video, so 'playback' will almost certainly be slower than > real-time. How do I control the pipeline so I process a single frame before > decoding the next ? > > I don't care about any audio track that might or might not be present. > > Any pointers or tips you can offer would be greatly appreciated :) > > Thanks in advance, > > Darren > > > ------------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From thomas at apestaart.org Tue Jan 6 13:43:51 2009 From: thomas at apestaart.org (Thomas Vander Stichele) Date: Tue, 06 Jan 2009 13:43:51 +0100 Subject: [gst-devel] gnonlin In-Reply-To: <1230478807.26858.0.camel@joder> References: <1230031442.3627.6.camel@otto.amantes> <1230478807.26858.0.camel@joder> Message-ID: <1231245831.5373.83.camel@ana.amantes> On Sun, 2008-12-28 at 16:40 +0100, Edward Hervey wrote: > Small update, > > The latest commit to gnonlin should have fixed some more issues if not > all. > > Edward Hi Edward, thanks for looking at this. However, even with the update, I still don't get 114 seconds of continuous audio with the mixes in the proper place using the test script I gave you. Are you sure that it works for you ? Happy newyear, Thomas -- I listen to the sounds they make on their way down I follow with my eyes till they crash -- Flumotion - the only way to stream! http://www.flumotion.net/ From ensonic at hora-obscura.de Tue Jan 6 14:15:26 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Tue, 06 Jan 2009 15:15:26 +0200 Subject: [gst-devel] appsrc and appsink elements In-Reply-To: <4961465A.4010302@gmail.com> References: <48903C57.2000904@st.com> <1217774661.23199.4.camel@metal> <4941D9C5.60909@gmail.com> <494215F7.7050908@livecliq.com> <4961465A.4010302@gmail.com> Message-ID: <4963596E.6050902@hora-obscura.de> Andres Colubri schrieb: > Hi, > > I've been struggling for a while with the appsrc element, and I wonder > if anyone has any ideas about this. Basically, I'm trying to capture the > contents of the screen into a video file. I'm not sure how to construct > the pipeline to do it. I know that it has to start with an appsrc and > end at a filesink, but I don't know how to initialize the appsrc (caps, > max-bytes, etc) and what other intermediate elements are needed > (encoders, demuxers). I looked at the examples that come with the source > of appsrc, but these are for different scenarios (reading data from a > file, streaming). Appsrc is for injecting data from applications. Appsrc & appsink allow to integrate gstreamer into a mulimedia application that uses a diffrent framework already. For screencapture you could use ximagesrc at the start of your pipeline. Stefan > > Any help will be greatly appreciated. > > Thanks, > Andres > > > Jure Zitnik wrote: >> Hi Andres, >> The appsrc and appsink are not part of the binary gstreamer-plugins-bad >> package on ubuntu 8.10 (reasoning is that they "still randomly change >> API and ABI" :-) >> But you can always build them from source: >> apt-get source gstreamer0.10-plugins-bad >> cd gst-plugins-bad0.10-0.10.5 (you might need to cange the version here) >> ./configure >> make >> >> After successful build, you can copy required shared libraries manually >> (if you just need appsrc/appsink) as follows: >> cp gst/app/.libs/libgstapp.so /usr/lib/gstreamer-0.10/ >> cp gst-libs/gst/app/.libs/libgstapp-0.10.so.0.0.0 /usr/local/lib >> You might need to create additional symbolic links in /usr/local/lib >> (both linking to libgstapp-0.10.so.0.0.0) : libgstapp-0.10.so.0 and >> libgstapp-0.10.so >> >> Now use gst-inspect to see if everything works fine... >> >> Jure >> >> Andres Colubri wrote: >> >>> Hello, >>> >>> I have tried to use appsrc, but for some reason it doesn't seem to be >>> recognized (I'm using gst-plugins-bad 0.10.8 on ubuntu 0.8.10) by >>> gstreamer. For example, when I run the appsrc_ex example, I get an >>> assertion error right after >>> app->src = gst_element_factory_make("appsrc", NULL); >>> so seems that app->src is null >>> >>> Also, gst-inspect doesn't find neither appsrc nor appsink. >>> >>> Thanks >>> Andres >>> >>> Wim Taymans wrote: >>> >>> >>>> On Wed, 2008-07-30 at 12:03 +0200, David SIORPAES wrote: >>>> >>>> >>>> >>>>> Hello, >>>>> >>>>> I would like to exchange buffers which are allocated and managed by an >>>>> application of mine to and from a GStreamer pipeline. >>>>> >>>>> By reading the documentation it looks like the appsrc/appsink are the >>>>> solution for me, but I am also reading here >>>>> http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/section-data-spoof.html >>>>> that >>>>> >>>>> "New API is being developed at the moment to make data insertion and >>>>> extraction less painful for applications. It can be found as GstAppSrc >>>>> and GstAppSink in the gst-plugins-bad module. At the time of writing >>>>> (October 2007), this API is not quite stable and ready yet, even though >>>>> it may work fine for your purposes." >>>>> >>>>> So, may I ask whether it is safe to use such elements or not ? Or are >>>>> there alternative solutions ? >>>>> >>>>> >>>>> >>>> The versionsof appsink/appsrc in the latest release of gst-plugins-bad >>>> are full-featured and ready for consumption. See >>>> http://webcvs.freedesktop.org/gstreamer/gst-plugins-bad/examples/app/ >>>> for a whole bunch of examples of different use cases. >>>> >>>> Wim >>>> >>>> >>>> >>>>> Thanks, >>>>> >>>>> David S. >>>>> >>>>> >>>>> >>>>> >>>>> ------------------------------------------------------------------------- >>>>> This SF.Net email is sponsored by the Moblin Your Move Developer's challenge >>>>> Build the coolest Linux based applications with Moblin SDK & win great prizes >>>>> Grand prize is a trip for two to an Open Source event anywhere in the world >>>>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>>>> _______________________________________________ >>>>> gstreamer-devel mailing list >>>>> gstreamer-devel at lists.sourceforge.net >>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>> >>>>> >>>>> >>>> ------------------------------------------------------------------------- >>>> This SF.Net email is sponsored by the Moblin Your Move Developer's challenge >>>> Build the coolest Linux based applications with Moblin SDK & win great prizes >>>> Grand prize is a trip for two to an Open Source event anywhere in the world >>>> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> gstreamer-devel at lists.sourceforge.net >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>>> >>>> >>>> >>> ------------------------------------------------------------------------------ >>> SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. >>> The future of the web can't happen without you. Join us at MIX09 to help >>> pave the way to the Next Web now. Learn more and register at >>> http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> >> ------------------------------------------------------------------------------ >> SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. >> The future of the web can't happen without you. Join us at MIX09 to help >> pave the way to the Next Web now. Learn more and register at >> http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From ensonic at hora-obscura.de Tue Jan 6 14:20:40 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Tue, 06 Jan 2009 15:20:40 +0200 Subject: [gst-devel] otem-gstreamer-video-thumbnailer sometimes fails to generate the thumbnail of ASF file In-Reply-To: References: Message-ID: <49635AA8.8090402@hora-obscura.de> hi, Lin Raffel schrieb: > I found totem-gstreamer-video-thumbnailer sometimes fails to generate > the thumbnail of ASF file, > the output message is. > > Reason: Took too much time to process. > > Is this the problem with the ASF file itself ? Its also happens with AVI files for me. IMHO its because for most files demuxers always scan the index and if that files has a very finegrained index it takes long. Its fixable, but requires quite some work. Stefan From ensonic at hora-obscura.de Tue Jan 6 14:25:04 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Tue, 06 Jan 2009 15:25:04 +0200 Subject: [gst-devel] VST Audio Plug in approach using Gstreamer and Linux OS with ARM Cortex A8 processors In-Reply-To: References: Message-ID: <49635BB0.5080002@hora-obscura.de> thx1138 schrieb: > Hello, > > We are presently using VST as a plug-in for audio applications and the > question came up today is it possible for VST plug-ins to work with > G-Streamer? Our VST's are migrating from PC/Mac Intel host and desire to use > VST approach with Gstreamer for ARM Cortex A8 processor. > > We have Linux Gstreamer as our Linux interface and we are wondering what has > been done in this area already to look at the API interfaces etc. If those are your own vst plugins I would port them to gstreamer. For vst effects you could use GstAudioFilter as a baseclass (gst-plugins-base/gst-libs/gst/audio/). If you want to keep the plugins as they are, you would need to write a brige plugin. It would load the vst plugins as a host an externaly appear as a gstreamer plugin. Stefan > > Regards, > > Terry > > > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From ensonic at hora-obscura.de Tue Jan 6 14:28:18 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Tue, 06 Jan 2009 15:28:18 +0200 Subject: [gst-devel] Hi In-Reply-To: <4961ACC8.6040008@nestgroup.net> References: <4961ACC8.6040008@nestgroup.net> Message-ID: <49635C72.2050104@hora-obscura.de> vaisakh.n schrieb: > help me to use gst-editor-0.8. You should use gst-editor from cvs. Its the ported version for 0.10. But be aware that it is incomplete. > I created xml file like this > > ./gst-launch-0.10 filesrc location=KDE_Logout_1.ogg ! oggdemux ! > vorbisdec ! audioconvert ! audioresample ! osssink -o test3.xml > > and > started the gst-editor like > ./gst-editor test3.xml Also be aware that gst-editor is limmited in what it can do. E.g. supporting demuxers (any elemnts with sometimes pads), does not make that much sense with the xml import/export. Stefan > > > but it shows error > like this > > /**********************************************/ > > > > ** Message: palette visible: TRUE > ** Message: making new palette > > (gst-editor:5188): GStreamer-CRITICAL **: file gstxml.c: line 415 > (gst_xml_make_element): assertion `element != NULL' failed > > (gst-editor:5188): GStreamer-CRITICAL **: file gstobject.c: line 223 > (gst_object_ref): assertion `GST_IS_OBJECT (object)' failed > > (gst-editor:5188): GStreamer-CRITICAL **: file gstobject.c: line 563 > (gst_object_unparent): assertion `object != NULL' failed > > (gst-editor:5188): GStreamer-CRITICAL **: file gstbin.c: line 520 > (gst_bin_add): assertion `GST_IS_ELEMENT (element)' failed > ** Message: linking GUI for vorbisdec0:src and audioconvert0:sink > ** Message: linking GUI for filesrc0:src and oggdemux0:sink > /************************************************/ > > and in gst-editor gui no proper links among the elements > > > help me... > thanks.. > > ***** Confidentiality Statement/Disclaimer ***** > > This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. > The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. > > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From ensonic at hora-obscura.de Tue Jan 6 14:32:31 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Tue, 06 Jan 2009 15:32:31 +0200 Subject: [gst-devel] 3gp demux plugin working on PC but not on target In-Reply-To: <89635EF2797FED4B9EF9769F80CEABC81AF3B1@EXGZ01.sasken.com> References: <89635EF2797FED4B9EF9769F80CEABC81AF3B1@EXGZ01.sasken.com> Message-ID: <49635D6F.5010800@hora-obscura.de> Rahul Verma schrieb: > > Hi, > > I am developing a 3gp Demux plugin. > > My pipeline is filesrc ! 3gpreader(custom) ! mpeg4decoder(custom) ! > xvimagesink. This works properly on the PC. But when I try to run the > same pipeline on target > > filesrc ! 3gpreader(custom) ! mpeg4decoder(custom) ! v4l2videosink, it > does not work. > > I get the following error : > > gst_bus_new: created new bus > > Caught SIGSEGV accessing address (nil) > > Spinning. Please run 'gdb gst-launch 129' to continue debugging, Ctrl-C > to quit, or Ctrl-\ to dump core. > > After checking the source files it is while attaching the bus to the > pipeline. > > However pipeline > filesrc ! ffdemux_mov_mp4_m4a_3gp_3g2_mj2 ! mpeg4decoder(custom) ! > v4l2videosink works fine with GST_DEBUG=0. > > In this pipeline also if I set GST_DEBUG=3/4/5 i get the same SIGSEGV > fault at the same point. > > I am stuck here and need some pointers. > > Regards, > Rahul We can't help you with such problems, as we don't have the sources for your elements. If you don't have a gdb or your tagets, you need to try to get some sort of logging working to track the problem down. Stefan From hadess at hadess.net Tue Jan 6 14:33:34 2009 From: hadess at hadess.net (Bastien Nocera) Date: Tue, 06 Jan 2009 13:33:34 +0000 Subject: [gst-devel] otem-gstreamer-video-thumbnailer sometimes fails to generate the thumbnail of ASF file In-Reply-To: <49635AA8.8090402@hora-obscura.de> References: <49635AA8.8090402@hora-obscura.de> Message-ID: <1231248814.3132.5663.camel@cookie.hadess.net> On Tue, 2009-01-06 at 15:20 +0200, Stefan Kost wrote: > hi, > > Lin Raffel schrieb: > > I found totem-gstreamer-video-thumbnailer sometimes fails to generate > > the thumbnail of ASF file, > > the output message is. > > > > Reason: Took too much time to process. > > > > Is this the problem with the ASF file itself ? > > Its also happens with AVI files for me. IMHO its because for most files demuxers > always scan the index and if that files has a very finegrained index it takes > long. Its fixable, but requires quite some work. As Stefan mentioned, it usually happens with broken files. You can use the "--no-limit" command-line option, but right now the thumbnailer limits itself to using 30 seconds of wall-clock time at the maximum, so as to avoid hanging indefinitely. If the file isn't broken, and plays straight away in Totem, you might have found a bug though. Cheers From ensonic at hora-obscura.de Tue Jan 6 14:46:39 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Tue, 06 Jan 2009 15:46:39 +0200 Subject: [gst-devel] otem-gstreamer-video-thumbnailer sometimes fails to generate the thumbnail of ASF file In-Reply-To: <1231248814.3132.5663.camel@cookie.hadess.net> References: <49635AA8.8090402@hora-obscura.de> <1231248814.3132.5663.camel@cookie.hadess.net> Message-ID: <496360BF.8080401@hora-obscura.de> Bastien Nocera schrieb: > On Tue, 2009-01-06 at 15:20 +0200, Stefan Kost wrote: >> hi, >> >> Lin Raffel schrieb: >>> I found totem-gstreamer-video-thumbnailer sometimes fails to generate >>> the thumbnail of ASF file, >>> the output message is. >>> >>> Reason: Took too much time to process. >>> >>> Is this the problem with the ASF file itself ? >> Its also happens with AVI files for me. IMHO its because for most files demuxers >> always scan the index and if that files has a very finegrained index it takes >> long. Its fixable, but requires quite some work. > > As Stefan mentioned, it usually happens with broken files. > > You can use the "--no-limit" command-line option, but right now the > thumbnailer limits itself to using 30 seconds of wall-clock time at the > maximum, so as to avoid hanging indefinitely. > > If the file isn't broken, and plays straight away in Totem, you might > have found a bug though. The files are actually valid, they are either very long or have insanely dense index entries. One easy way to check if that is the case if filesrc ! avidemux ! ... va. filesrc ! queue ! avidemux The later switches to streaming mode where the index is not scanned. One way to solve this could be to build the index while playing. Then one can easily seek back. If one seeks forward, the demuxer would scan. The current behaviour of pre-scanning the index in not very smart as people do not always seek. There might be some issues with such a change as well. From looking at avidemux: - the reported duration is adjusted to the duration from index if it diferes from the values in the header - the index is sorted, in theory entries in a avi might come in a random order (can they?) Stefan > > Cheers > > > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From liangzhihong1984 at 126.com Tue Jan 6 14:51:27 2009 From: liangzhihong1984 at 126.com (liangzhihong1984) Date: Tue, 6 Jan 2009 21:51:27 +0800 (CST) Subject: [gst-devel] How to implement the pipeline in programming? Message-ID: <33108195.807721231249887236.JavaMail.coremail@bj126app55.126.com> Hi, everyone. I just got a pipeline like this: gst-launch -v gstrtpbin name=rtpbin filesrc location=h264.ts ! \ queue2 max-size-buffers=65535 ! \ h264parse ! \ rtpmp2tpay ! \ queue2 max-size-buffers=65535 ! \ identity sync=true silent=true sleep-time=300 ! \ rtpbin.send_rtp_sink_0 \ rtpbin.send_rtp_src_0 ! udpsink host=192.168.1.2 port=5000 \ rtpbin.send_rtcp_src_0 ! udpsink host=192.168.1.2 port=5001 sync=false async=false \ udpsrc port=5002 ! rtpbin.recv_rtcp_sink_0 Now, I want to do the same work with programming language, such as C or C++. But I've no idea about how to do that, especially how to handle these rtpbin related stuffs, like "send_rtp_src_0" and "send_rtcp_src_0". How to create and link these elements? Can anybody give me some help. Thank you in advance. Best Regards, Leon Liang -------------- next part -------------- An HTML attachment was scrubbed... URL: From jack.foster11 at gmail.com Tue Jan 6 15:20:12 2009 From: jack.foster11 at gmail.com (Jack Foster) Date: Tue, 6 Jan 2009 09:20:12 -0500 Subject: [gst-devel] How to implement the pipeline in programming? In-Reply-To: <33108195.807721231249887236.JavaMail.coremail@bj126app55.126.com> References: <33108195.807721231249887236.JavaMail.coremail@bj126app55.126.com> Message-ID: lol. I asked this exact question not five days ago. Check this answer from Wim: I added some C examples in CVS here: http://webcvs.freedesktop.org/gstreamer/gst-plugins-good/tests/examples/rtp/ Look at client-PCMA.c and server-alsasrc-PCMA.c Cheers! Jack. On Tue, Jan 6, 2009 at 8:51 AM, liangzhihong1984 wrote: > > Hi, everyone. > I just got a pipeline like this: > > > gst-launch -v gstrtpbin name=*rtpbin* filesrc location=h264.ts ! \ > > queue2 max-size-buffers=65535 ! \ > > h264parse ! \ > > rtpmp2tpay ! \ > > queue2 max-size-buffers=65535 ! \ > > identity sync=true silent=true sleep-time=300 ! \ > > *rtpbin*.send_rtp_sink_0 \ > > *rtpbin*.send_rtp_src_0 ! udpsink host=192.168.1.2 > port=5000 \ > > *rtpbin*.send_rtcp_src_0 ! udpsink host=192.168.1.2 > port=5001 sync=false async=false \ > > udpsrc port=5002 ! *rtpbin*.recv_rtcp_sink_0 > > > > Now, I want to do the same work with programming language, such as C or > C++. > > But I've no idea about how to do that, especially how to handle these > rtpbin related > > stuffs, like "send_rtp_src_0" and "send_rtcp_src_0". > > How to create and link these elements? > > Can anybody give me some help. > > Thank you in advance. > > > > > > > > Best Regards, > > Leon Liang > > > ------------------------------ > ???????????????????5000???? > > ------------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jack.foster11 at gmail.com Tue Jan 6 16:04:17 2009 From: jack.foster11 at gmail.com (Jack Foster) Date: Tue, 6 Jan 2009 10:04:17 -0500 Subject: [gst-devel] Crackling/Popping Message-ID: I get a significant amount of popping and crackling with these examples. Is this a buffering/latency issues? I'm not sure how it could be as I'm running this demo on my local machine. Jevin On Fri, Jan 2, 2009 at 10:22 AM, Wim Taymans wrote: > On Wed, 2008-12-31 at 11:03 -0500, Jack Foster wrote: > > Hi gang! > > > > I see lots of examples that have CLI to gStreamer but surely people > > are building applications with C. It seems like an awesome framework > > but I'm just trying to get my head around it. I would like to see > > more C examples. > > > > In any case, I'm trying to a simple audio rtp sender/receiver working > > but am having a hard time translating the example from the gstrtpbin > > documentation to C. I'll then post a working copy on the mailing list > > for others running into the same issue. > > > > I really have no idea if the errors are related to the mad plugin or > > the pads not linked together properly. > > > > Attached is the error and the program. Any help would be greatly > > appreciated. > > I added some C examples in CVS here: > > > http://webcvs.freedesktop.org/gstreamer/gst-plugins-good/tests/examples/rtp/ > > Look at client-PCMA.c and server-alsasrc-PCMA.c > > Wim > > > > > Thanks! > > Jack > > > ------------------------------------------------------------------------------ > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From wim.taymans at gmail.com Tue Jan 6 16:22:22 2009 From: wim.taymans at gmail.com (Wim Taymans) Date: Tue, 6 Jan 2009 16:22:22 +0100 Subject: [gst-devel] Crackling/Popping In-Reply-To: References: Message-ID: It's the clock slaving that causes this, our current algorithm is rather crude and just skips 10ms of samples or inserts silence to keep up with the different clock rates. Even if you are running on a local machine, the sender side will use the system clock while the receiver will ultimately use the audio clock for rendering the samples. Wim On Tue, Jan 6, 2009 at 4:04 PM, Jack Foster wrote: > I get a significant amount of popping and crackling with these examples. Is > this a buffering/latency issues? I'm not sure how it could be as I'm > running this demo on my local machine. > > Jevin > > On Fri, Jan 2, 2009 at 10:22 AM, Wim Taymans wrote: >> >> On Wed, 2008-12-31 at 11:03 -0500, Jack Foster wrote: >> > Hi gang! >> > >> > I see lots of examples that have CLI to gStreamer but surely people >> > are building applications with C. It seems like an awesome framework >> > but I'm just trying to get my head around it. I would like to see >> > more C examples. >> > >> > In any case, I'm trying to a simple audio rtp sender/receiver working >> > but am having a hard time translating the example from the gstrtpbin >> > documentation to C. I'll then post a working copy on the mailing list >> > for others running into the same issue. >> > >> > I really have no idea if the errors are related to the mad plugin or >> > the pads not linked together properly. >> > >> > Attached is the error and the program. Any help would be greatly >> > appreciated. >> >> I added some C examples in CVS here: >> >> >> http://webcvs.freedesktop.org/gstreamer/gst-plugins-good/tests/examples/rtp/ >> >> Look at client-PCMA.c and server-alsasrc-PCMA.c >> >> Wim >> >> > >> > Thanks! >> > Jack >> > >> > ------------------------------------------------------------------------------ >> > _______________________________________________ >> > gstreamer-devel mailing list >> > gstreamer-devel at lists.sourceforge.net >> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> ------------------------------------------------------------------------------ >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From andres.colubri at gmail.com Tue Jan 6 16:33:44 2009 From: andres.colubri at gmail.com (Andres Colubri) Date: Tue, 06 Jan 2009 07:33:44 -0800 Subject: [gst-devel] appsrc and appsink elements In-Reply-To: <4963596E.6050902@hora-obscura.de> References: <48903C57.2000904@st.com> <1217774661.23199.4.camel@metal> <4941D9C5.60909@gmail.com> <494215F7.7050908@livecliq.com> <4961465A.4010302@gmail.com> <4963596E.6050902@hora-obscura.de> Message-ID: <496379D8.7070600@gmail.com> >> I've been struggling for a while with the appsrc element, and I wonder >> if anyone has any ideas about this. Basically, I'm trying to capture the >> contents of the screen into a video file. I'm not sure how to construct >> the pipeline to do it. I know that it has to start with an appsrc and >> end at a filesink, but I don't know how to initialize the appsrc (caps, >> max-bytes, etc) and what other intermediate elements are needed >> (encoders, demuxers). I looked at the examples that come with the source >> of appsrc, but these are for different scenarios (reading data from a >> file, streaming). >> > > Appsrc is for injecting data from applications. Appsrc & appsink allow to > integrate gstreamer into a mulimedia application that uses a diffrent framework > already. > For screencapture you could use ximagesrc at the start of your pipeline. > Thanks for your answer, Stefan. I realize that I didn't specify my problem very clearly, because even though the end result of the pipeline consists in capturing the content of the application window (not the whole screen), I need a solution that is crossplatform. I did some tests already, and I can grab the contents of the window into memory buffers, so it seems to me that the best approach would be to push these buffers through an appsrc element. Regards, Andres From wingo at pobox.com Tue Jan 6 17:51:38 2009 From: wingo at pobox.com (Andy Wingo) Date: Tue, 06 Jan 2009 17:51:38 +0100 Subject: [gst-devel] Conversion to GIT (no, really, I mean it this time) In-Reply-To: <1231239971.9864.24.camel@joder> (Edward Hervey's message of "Tue, 06 Jan 2009 12:06:11 +0100") References: <1231239971.9864.24.camel@joder> Message-ID: On Tue 06 Jan 2009 12:06, Edward Hervey writes: > ... WE ARE REALLY SWITCHING TO GIT THIS MONTH ! Hoooooooooooorraaaaaaaaaaaaaaaaaaaaaaaaaay !!!! One One One ! > * Go party Yes sir. Andy -- http://wingolog.org/ From msmith at xiph.org Tue Jan 6 20:08:22 2009 From: msmith at xiph.org (Michael Smith) Date: Tue, 6 Jan 2009 11:08:22 -0800 Subject: [gst-devel] 3gp demux plugin working on PC but not on target In-Reply-To: <89635EF2797FED4B9EF9769F80CEABC81AF3B1@EXGZ01.sasken.com> References: <89635EF2797FED4B9EF9769F80CEABC81AF3B1@EXGZ01.sasken.com> Message-ID: <3c1737210901061108x6fe903d5w7ccff230ce21dba@mail.gmail.com> On Tue, Jan 6, 2009 at 2:40 AM, Rahul Verma wrote: > > Hi, > > I am developing a 3gp Demux plugin. > > My pipeline is filesrc ! 3gpreader(custom) ! mpeg4decoder(custom) ! > xvimagesink. This works properly on the PC. But when I try to run the same > pipeline on target > > filesrc ! 3gpreader(custom) ! mpeg4decoder(custom) ! v4l2videosink, it does > not work. > > I get the following error : > > gst_bus_new: [00m created new bus > > Caught SIGSEGV accessing address (nil) Without the source to your plugin, all we can say about this particular crash is "it's presumably a bug in your plugin, you'll need to fix it". That said, why aren't you using qtdemux? GStreamer includes a fairly full-featured, high quality demuxer for quicktime-and-qt-derived-formats like 3gp, and it works well already. Mike From ensonic at hora-obscura.de Tue Jan 6 21:31:16 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Tue, 06 Jan 2009 22:31:16 +0200 Subject: [gst-devel] Conversion to GIT (no, really, I mean it this time) In-Reply-To: <1231239971.9864.24.camel@joder> References: <1231239971.9864.24.camel@joder> Message-ID: <4963BF94.9080503@hora-obscura.de> Edward Hervey schrieb: > Hi all, > > No, this is no longer another boring mail from myself regarding test > repositories and all that crap... > > ... WE ARE REALLY SWITCHING TO GIT THIS MONTH ! > > The current switch date is set to just after the upcoming releases of > core/base/python/bad, which should be on Monday 19th of January. > Do we have the "simple work flow docs for GStreamer developers" as mentioned on http://gstreamer.freedesktop.org/wiki/GitMigration Thanks for making it happen && happy new year! Stefan > > What will happen at that point: > * All of gstreamer cvs repositories will become read-only > * All the gstreamer modules will be converted to git using the scripts > used to do the previous git test conversions > * The git repositories will be pushed as read-only to the main > freedesktop.org servers. > * After some final integrity checks (hopefully less than a few hours) > the repositories will be made read/write > * Go party > * Make GStreamer even more awesome > > A big thanks to everyone involved for their patience regarding this > conversion, it's been a hectic road since the decision back in Istanbul, > but I think it was worth the ride. > > Edward > > > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From thaytan at noraisin.net Tue Jan 6 23:02:06 2009 From: thaytan at noraisin.net (Jan Schmidt) Date: Tue, 06 Jan 2009 22:02:06 +0000 Subject: [gst-devel] Releases of all of Core/Base/Bad/Python this month! All 4 are now frozen. Message-ID: <1231279326.6539.5.camel@fancy-ubuntu> Hi all, All 4 of the Core, Base Plugins, Bad Plugins and Python bindings modules are now frozen for releases. The Bad Plugins aren't normally in this cycle. They're there this month to handle appsrc and appsink moving from -bad to -base at last, and audioresample trading places with the much better speexresample. Speexresample is now in -base, but registers with the name 'audioresample'. The old audioresample is now in -bad as 'legacyresample' See the release plan for the first few months of this year (http://www.gstreamer.net/wiki/ReleasePlanning2009-1) for more detail, and expect the pre-releases later tonight. Cheers, Jan. -- Jan Schmidt From thaytan at noraisin.net Wed Jan 7 01:01:11 2009 From: thaytan at noraisin.net (Jan Schmidt) Date: Wed, 07 Jan 2009 00:01:11 +0000 Subject: [gst-devel] Pre-releases of Core/Base/Bad/Python available In-Reply-To: <1231279326.6539.5.camel@fancy-ubuntu> References: <1231279326.6539.5.camel@fancy-ubuntu> Message-ID: <1231286471.6539.17.camel@fancy-ubuntu> Hi all, The first pre-releases of the 4 modules are now available at: http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.21.2.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.21.2.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-bad/pre/gst-plugins-bad-0.10.9.2.tar.bz2 and http://gstreamer.freedesktop.org/src/gst-python/pre/gst-python-0.10.13.2.tar.bz2 Please test, and file any bugs at http://bugzilla.gnome.org/ ! Expect new pre-releases on Friday if needed. Check the release plan to find out about the release schedule: http://www.gstreamer.net/wiki/ReleasePlanning2009-1 Cheers, Jan. On Tue, 2009-01-06 at 22:02 +0000, Jan Schmidt wrote: > Hi all, > > All 4 of the Core, Base Plugins, Bad Plugins and Python bindings modules > are now frozen for releases. The Bad Plugins aren't normally in this > cycle. They're there this month to handle appsrc and appsink moving from > -bad to -base at last, and audioresample trading places with the much > better speexresample. Speexresample is now in -base, but registers with > the name 'audioresample'. The old audioresample is now in -bad as > 'legacyresample' > > See the release plan for the first few months of this year > (http://www.gstreamer.net/wiki/ReleasePlanning2009-1) for more detail, > and expect the pre-releases later tonight. > > Cheers, > Jan. -- Jan Schmidt From ds at entropywave.com Wed Jan 7 02:38:58 2009 From: ds at entropywave.com (David Schleef) Date: Tue, 6 Jan 2009 17:38:58 -0800 Subject: [gst-devel] [gst-daily] How to create raw video In-Reply-To: References: Message-ID: <20090107013858.GA4332@entropywave.com> On Thu, Jan 01, 2009 at 06:18:27PM -0800, Adam B wrote: > Hello all, > > I have a bunch of TGA image files that I want to convert into a raw, > uncompressed video. Can I do this with gst-launch? Seems like it would be > a simple thing to do but I'm not having much luck with my google searches. It's much simpler if you have PNG images, in which case you can start from the pipeline in the wiki page: http://gstreamer.freedesktop.org/wiki/MultiFileSrc dave... From dstaples at friendmts.co.uk Wed Jan 7 12:06:26 2009 From: dstaples at friendmts.co.uk (Darren Staples) Date: Wed, 07 Jan 2009 11:06:26 +0000 Subject: [gst-devel] FLV demuxing and decoding Message-ID: Hi again, As an experiment I?ve been trying to get flash video (.flv) files to play in an x-window. I can get a video-test signal to display using: ?gst-launch-0.10 videotestsrc ! Ximagesink?, so I know the basic infra-structure is working :) As for getting .flv files to play I?ve tried variations on: ?gst-launch-0.10 filesrc location=file.flv ! flv_demux ! ffdec_flv ! ximagesink?. But I keep getting an internal data stream error at the flvdemux element. I have the base, bad and FFMPEG plugins installed. Is this basic pipeline correct or am I doing something stupid ? Thanks in advance ! Darren *** FRIEND MTS LIMITED EMAIL CONFIDENTIALITY *** This email message and any attachments may contain information which is confidential or privileged and is intended for the sole use of the person to whom it is addressed. If you are not the intended recipient, be aware that any disclosures, copying, distribution or use of the contents is prohibited. If you have received this email message in error, please notify our office by telephone (+44 (0)121 633 2000) or email (admin at friendmts.co.uk) immediately. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilboed at gmail.com Wed Jan 7 12:40:05 2009 From: bilboed at gmail.com (Edward Hervey) Date: Wed, 07 Jan 2009 12:40:05 +0100 Subject: [gst-devel] FLV demuxing and decoding In-Reply-To: References: Message-ID: <1231328405.17445.0.camel@localhost> Maybe because the files you are trying don't contain FLV1 video but some other codec ? replace flvdemux by decodebin2, it'll do all the magic for you Edward On Wed, 2009-01-07 at 11:06 +0000, Darren Staples wrote: > Hi again, > > As an experiment I?ve been trying to get flash video (.flv) files to > play in an x-window. > I can get a video-test signal to display using: ?gst-launch-0.10 > videotestsrc ! Ximagesink?, so I know the basic infra-structure is > working :) > > As for getting .flv files to play I?ve tried variations on: > ?gst-launch-0.10 filesrc location=file.flv ! flv_demux ! ffdec_flv ! > ximagesink?. > But I keep getting an internal data stream error at the flvdemux > element. > > I have the base, bad and FFMPEG plugins installed. > > Is this basic pipeline correct or am I doing something stupid ? > > Thanks in advance ! > > Darren > > > > > *** FRIEND MTS LIMITED EMAIL CONFIDENTIALITY *** > > This email message and any attachments may contain information which > is confidential or privileged and is intended for the sole use of the > person to whom it is addressed. If you are not the intended > recipient, be aware that any disclosures, copying, distribution or use > of the contents is prohibited. If you have received this email > message in error, please notify our office by telephone (+44 (0)121 > 633 2000) or email (admin at friendmts.co.uk) immediately. Thank you. > > ------------------------------------------------------------------------------ > Check out the new SourceForge.net Marketplace. > It is the best place to buy or sell services for > just about anything Open Source. > http://p.sf.net/sfu/Xq1LFB > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From bisht.sudarshan at gmail.com Wed Jan 7 12:39:21 2009 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Wed, 7 Jan 2009 17:09:21 +0530 Subject: [gst-devel] FLV demuxing and decoding In-Reply-To: References: Message-ID: <785339900901070339q75455f3fq737b450289c07069@mail.gmail.com> Hi ,, Use gst-launch filesrc location=file.flv ! decodebin ! ffmpegcolorspace ! ximagesink , here decodebin automatically links the required plugins . On Wed, Jan 7, 2009 at 4:36 PM, Darren Staples wrote: > Hi again, > > As an experiment I've been trying to get flash video (.flv) files to play > in an x-window. > I can get a video-test signal to display using: "gst-launch-0.10 > videotestsrc ! Ximagesink", so I know the basic infra-structure is working > :) > > As for getting .flv files to play I've tried variations on: > "gst-launch-0.10 filesrc location=file.flv ! flv_demux ! ffdec_flv ! > ximagesink". > But I keep getting an internal data stream error at the flvdemux element. > > I have the base, bad and FFMPEG plugins installed. > > Is this basic pipeline correct or am I doing something stupid ? > > Thanks in advance ! > > Darren > > > > *** FRIEND MTS LIMITED EMAIL CONFIDENTIALITY *** > > This email message and any attachments may contain information which is > confidential or privileged and is intended for the sole use of the person to > whom it is addressed. If you are not the intended recipient, be aware that > any disclosures, copying, distribution or use of the contents is > prohibited. If you have received this email message in error, please notify > our office by telephone (+44 (0)121 633 2000) or email ( > admin at friendmts.co.uk) immediately. Thank you. > > > ------------------------------------------------------------------------------ > Check out the new SourceForge.net Marketplace. > It is the best place to buy or sell services for > just about anything Open Source. > http://p.sf.net/sfu/Xq1LFB > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From rahul.verma at sasken.com Wed Jan 7 13:02:43 2009 From: rahul.verma at sasken.com (Rahul Verma) Date: Wed, 7 Jan 2009 17:32:43 +0530 Subject: [gst-devel] 3gp demux plugin working on PC but not on target Message-ID: <89635EF2797FED4B9EF9769F80CEABC81AF3B7@EXGZ01.sasken.com> Hi, > That said, why aren't you using qtdemux? GStreamer includes a fairly > full-featured, high quality demuxer for > quicktime-and-qt-derived-formats like 3gp, and it works well already. We use the qtdemux extensively and find it very stable. We are developing out 3gp plugin as an experiment just to learn more about the GStreamer framework. > Without the source to your plugin, all we can say about this > particular crash is "it's presumably a bug in your plugin, you'll need > to fix it". I will try to explain about my problem in more detail. here is the initial code snippet GST_BOILERPLATE(Gst3gpReader, gst_3gpreader, GstElement, GST_TYPE_ELEMENT); static gboolean plugin_init(GstPlugin *plugin) { GST_DEBUG_CATEGORY_INIT(gst_3gpreader_debug, "SASKEN3GPREADER", 0, "Sasken 3gp reader"); return gst_element_register(plugin, "sasken3gpreader", GST_RANK_NONE, GST_TYPE_3GPREADER); } static GstElementDetails element_details = { "sasken 3gp reader", "3gpreader", "sasken 3gp reader reader for GStreamer", "Rahul Verma" }; GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, "Sasken3gpReader", "Sasken 3gp reader plugin for GStreamer", plugin_init, VERSION, "Proprietary", "Sasken", "http://www.sasken.com") static void gst_3gpreader_base_init(gpointer klass) { //here I Just add the static pad templates to the element and the element details } static void gst_3gpreader_class_init(Gst3gpReaderClass *klass) { //here I just assign the change_state function pointer } static void gst_3gpreader_init(Gst3gpReader *filter, Gst3gpReaderClass *klass) { //here I set the activate, activate_push and activate_pull function pointer //add the sink pad } Then in the other functions I do the internal parsing and passing the buffer to the downstream plugins to be decoded and displayed. All this work fine in the host(PC). But on target it fails and i don't know how to proceed further. I am also attaching a part of the logs(related to 3gp reader because the entire logs was of 1.7 MB) Also the pipeline filesrc location=c.3gp ! ffdemux_mov_mp4_m4a_3gp_3g2_mj2 ! ffdec_mpeg4 ! v4l2videosink, with GST_DEBUG=3, gives the same problem, but works fine with GST_DEBUG=0. My pipeline doesn't work for any GST_DEBUG level. Any pointers will be helpful. Regards, Rahul > > Hi, > > I am developing a 3gp Demux plugin. > > My pipeline is filesrc ! 3gpreader(custom) ! mpeg4decoder(custom) ! > xvimagesink. This works properly on the PC. But when I try to run the same > pipeline on target > > filesrc ! 3gpreader(custom) ! mpeg4decoder(custom) ! v4l2videosink, it does > not work. > > I get the following error : > > gst_bus_new: [00m created new bus > > Caught SIGSEGV accessing address (nil) Without the source to your plugin, all we can say about this particular crash is "it's presumably a bug in your plugin, you'll need to fix it". That said, why aren't you using qtdemux? GStreamer includes a fairly full-featured, high quality demuxer for quicktime-and-qt-derived-formats like 3gp, and it works well already. Mike SASKEN BUSINESS DISCLAIMER ------------------------- This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, Disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: logs_target1.txt URL: From kumarkm at tataelxsi.co.in Wed Jan 7 12:56:45 2009 From: kumarkm at tataelxsi.co.in (Kumar) Date: Wed, 7 Jan 2009 17:26:45 +0530 Subject: [gst-devel] G726 RTP payload packetizer Message-ID: <011f01c970bf$041db510$04000100@telxsi.com> Hi all, To add one more point, is this implementation covers the recommendations of RFC 3551.? Thanks, Kumar KM -----Original Message----- From: Kumar [mailto:kumarkm at tataelxsi.co.in] Sent: Wednesday, January 07, 2009 5:12 PM To: Gstreamer-Devel (E-mail) Cc: 'dev-gstreamer at axis.com' Subject: G726 RTP payload packetizer Hi all, I have been using the G726 RTP packetizer for streaming from Gstreamer to VLC player. when I use a Little -endian(LE) input data, for 32kbps & 16kbps it works fine, without any noise. But for 24kbps & 40 kbps it gives noise for LE data. When I use all test streams with BE, then it works fine. Does this G726 RTP packetizer accept data in Big-endian(BE)? Can somebody give more info on this. Thanks in advance! Regards, Kumar KM The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From souilem_fadi at yahoo.fr Wed Jan 7 13:29:14 2009 From: souilem_fadi at yahoo.fr (souilem fadi) Date: Wed, 7 Jan 2009 12:29:14 +0000 (GMT) Subject: [gst-devel] Tr : Fwd: Let people know what happens there in PALESTINE >>> Message-ID: <103383.79008.qm@web24404.mail.ird.yahoo.com> I know it?may not be the appropriate?ML?, but believe?me when i received this mail from an?arabic ?friend , i finally found?that? all we heard? here ?on TV about Isreal fighting terrorists in?gaza strip?was just lies. i do execuse , but i don't appologize for forwarding this mail? ? SOUILEM FADI IT engineer, Voice&video over IP developer ----- Message transf?r? ---- De : ? SAMY MABROUK ? Envoy? le : Lundi, 5 Janvier 2009, 14h16mn 27s Objet?: Fwd: Let people know what happens there in PALESTINE >>> ---------- Forwarded message ---------- ? Dear friends and colleagues; ? The following is self-explanatory and talks for itself! ? Plzzzzz Keep forwarding this e-mail. Let people know what happens to us in Palestine. Por favor reenvia ese mail. El mundo debe saber que pasa alla. ????? ????? ?? ????? ??? ???????? ???? ?????? ???? ???? ???? ???? ?? ?????? ? Making sure they get to school. ? ? ? Helping Ladies across the street.. ? ? ? ? Providing childcare. ? ? ? Allowing them a place to rest (permanently) ? ? ? ? ? Access to Health care. ? ? ? ? Construction projects (demolition) ? ? ? ? ? Respecting American and British pacifist resisters (such as American Rachel Corrie) ? ? ? ? ? And others. ? ? ? ? And if you are not satisfied, now, with the truth the following pictures are war crimes as defined by the UN, The Hague and the Geneva Convention. ? ? Using images of your enemy dead or alive (violation) ? ? ? ? Human shields (violation) ? ? ? ? Live Burial Torture (violation) ? ? ? ? And as a last resort, Execution (violation) ? ? ? ?? These IDF soldiers have faces... I can clearly see them...Cant you? Why are they not being prosecuted? Because it is systematic process that is driven by the government designed to force the people of Palestine into exile so Israel can claim all the land and resources. This where my American tax dollars are going, do you know where your tax dollars are at? TAKE THE TIME TO FIND THE TRUTH. So many lives depend on it I, like so many Americans, am Caucasian, non-Arab, and religious. I can no longer sit back with good conscience and do nothing while my government is supporting the types of terrorist actions that we have condemned Muslim Fundamentalist for. Call your Congressman and Senator, send an email to the White House and demand that our government negotiate FAIRLY with both sides and bring a fair and just solution to Palestine and Israel . ? (CRUELTY OF ISRAEL PLS SEND IT TO YOUR FRIENDS ? ________________________________ Life on your PC is safer, easier, and more enjoyable with Windows Vista?. See how ________________________________ Glissez - d?placez : c'est si simple de partager vos photos sur Windows Live? Photos ! ________________________________ D?couvrez tout ce que Windows Live a ? vous apporter ! ________________________________ Tous vos amis discutent sur Messenger, et vous ? T?l?chargez Messenger, c'est gratuit ! ________________________________ Organisez une f?te en un tour de main : invitation, photos, blog? ________________________________ Tous vos amis discutent sur Messenger, et vous ? T?l?chargez Messenger, c'est gratuit ! --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "RT5_insat" group. To post to this group, send email to RT5_insat at googlegroups.com For more options, visit this group at http://groups.google.com/group/RT5_insat -~----------~----~----~----~------~----~------~--~--- -- "Wisdom lies not in obtaining knowledge, but in using it in the right way" - kroesjnov ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ MABROUK Samy Security Analyst at National Agency for Computer Security Computer Emergency Response Team CERT-TCC Professional mail: samy.mabrouk at ansi.tn -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 13481 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 25569 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... 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Name: not available Type: image/jpeg Size: 176748 bytes Desc: not available URL: From kumarkm at tataelxsi.co.in Wed Jan 7 12:41:52 2009 From: kumarkm at tataelxsi.co.in (Kumar) Date: Wed, 7 Jan 2009 17:11:52 +0530 Subject: [gst-devel] G726 RTP payload packetizer Message-ID: <011001c970bc$f06e8ff0$04000100@telxsi.com> Hi all, I have been using the G726 RTP packetizer for streaming from Gstreamer to VLC player. when I use a Little -endian(LE) input data, for 32kbps & 16kbps it works fine, without any noise. But for 24kbps & 40 kbps it gives noise for LE data. When I use all test streams with BE, then it works fine. Does this G726 RTP packetizer accept data in Big-endian(BE)? Can somebody give more info on this. Thanks in advance! Regards, Kumar KM The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From ylatuya at gmail.com Wed Jan 7 12:42:53 2009 From: ylatuya at gmail.com (Andoni Morales) Date: Wed, 7 Jan 2009 12:42:53 +0100 Subject: [gst-devel] New GStreamer binaries for Windows based on OAbuild. Message-ID: <772db3280901070342sfd0de76j9ca1169e29116972@mail.gmail.com> During the last couple of months we have been working on porting GStreamer to Windows using the OAbuild project. We introduced some modifications to OAbuild in order to improve compatibility and provide video decoding and encoding in a larger number of codecs. Changelog: ?Compile ffmpeg with mingw/msys with full codec support. ?Remove the ffmpeg project from OAbuild and link the ffmpeg plugin to the mingw/msys ffmpeg libraries. ?Remove the GLib project from OAbuild and link GStreamer to the GLib version provided with the GTK+ win32 installer (in this way GStreamer can be used by GTK applications). ?Update liboil to a newer version to compile the volume plugin. ?Change the ffmpeg plugin code to compile it using VS. ?Change the qtdemux plugin code to compile it using VS. ?Compile libmad for Windows and add the libraries to the Gstreamer's dependencies to provide mp3 decoding support. ?Add the mad plugin to OAbuild. ?Add the asfdemux plugin to OAbuild. ?Remove the Direct Show wrapper and use the ffmepg plugin for all the decoding stuff. ?Add the xvid plugin to OAbuild. ?Change the xvid plugin code to complile it usin VS. ?Add the mpeg2decoder plugin to OAbuild. ?Add the mpegstream plugin to OAbuild. To do: ?Make the x264 plugin work ?Improve mpeg2 decoding/demuxing support ?Add dvd navigation support You can find the installers at: http://forja.rediris.es/frs/?group_id=96 Andoni Morales Alastruey Andr?s Colubri -------------- next part -------------- An HTML attachment was scrubbed... URL: From julien.isorce at gmail.com Wed Jan 7 16:56:41 2009 From: julien.isorce at gmail.com (Julien Isorce) Date: Wed, 7 Jan 2009 16:56:41 +0100 Subject: [gst-devel] New GStreamer binaries for Windows based on OAbuild. In-Reply-To: <772db3280901070342sfd0de76j9ca1169e29116972@mail.gmail.com> References: <772db3280901070342sfd0de76j9ca1169e29116972@mail.gmail.com> Message-ID: <180a127d0901070756w23217a3dp7a7386880b047a7b@mail.gmail.com> Hi Nice, could you plan to add opengl plugin in the next release ? http://diracvideo.org/git?p=gst-plugins-gl.git;a=summary Sincerely Julien 2009/1/7 Andoni Morales > During the last couple of months we have been working on porting GStreamer > to Windows using the OAbuild project. We introduced some modifications to > OAbuild in order to improve compatibility and provide video decoding and > encoding in a larger number of codecs. > > Changelog: > ?Compile ffmpeg with mingw/msys with full codec support. > ?Remove the ffmpeg project from OAbuild and link the ffmpeg plugin to the > mingw/msys ffmpeg libraries. > ?Remove the GLib project from OAbuild and link GStreamer to the GLib > version provided > with the GTK+ win32 installer (in this way GStreamer can be used by GTK > applications). > ?Update liboil to a newer version to compile the volume plugin. > ?Change the ffmpeg plugin code to compile it using VS. > ?Change the qtdemux plugin code to compile it using VS. > ?Compile libmad for Windows and add the libraries to the Gstreamer's > dependencies to provide mp3 decoding support. > ?Add the mad plugin to OAbuild. > ?Add the asfdemux plugin to OAbuild. > ?Remove the Direct Show wrapper and use the ffmepg plugin for all the > decoding stuff. > ?Add the xvid plugin to OAbuild. > ?Change the xvid plugin code to complile it usin VS. > ?Add the mpeg2decoder plugin to OAbuild. > ?Add the mpegstream plugin to OAbuild. > > To do: > ?Make the x264 plugin work > ?Improve mpeg2 decoding/demuxing support > ?Add dvd navigation support > > > You can find the installers at: > http://forja.rediris.es/frs/?group_id=96 > > > Andoni Morales Alastruey > Andr?s Colubri > > > ------------------------------------------------------------------------------ > Check out the new SourceForge.net Marketplace. > It is the best place to buy or sell services for > just about anything Open Source. > http://p.sf.net/sfu/Xq1LFB > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ylatuya at gmail.com Wed Jan 7 17:36:15 2009 From: ylatuya at gmail.com (Andoni Morales) Date: Wed, 7 Jan 2009 17:36:15 +0100 Subject: [gst-devel] New GStreamer binaries for Windows based on OAbuild. In-Reply-To: <180a127d0901070756w23217a3dp7a7386880b047a7b@mail.gmail.com> References: <772db3280901070342sfd0de76j9ca1169e29116972@mail.gmail.com> <180a127d0901070756w23217a3dp7a7386880b047a7b@mail.gmail.com> Message-ID: <772db3280901070836x2157d230tec3c731932140972@mail.gmail.com> Sure! We are planning to create a project in sourceforge to encourage every to work on adding new plugins. Regards Andoni Morales 2009/1/7 Julien Isorce > Hi > > Nice, could you plan to add opengl plugin in the next release ? > > http://diracvideo.org/git?p=gst-plugins-gl.git;a=summary > > Sincerely > > Julien > > 2009/1/7 Andoni Morales > >> During the last couple of months we have been working on porting GStreamer >> to Windows using the OAbuild project. We introduced some modifications to >> OAbuild in order to improve compatibility and provide video decoding and >> encoding in a larger number of codecs. >> >> Changelog: >> ?Compile ffmpeg with mingw/msys with full codec support. >> ?Remove the ffmpeg project from OAbuild and link the ffmpeg plugin to the >> mingw/msys ffmpeg libraries. >> ?Remove the GLib project from OAbuild and link GStreamer to the GLib >> version provided >> with the GTK+ win32 installer (in this way GStreamer can be used by GTK >> applications). >> ?Update liboil to a newer version to compile the volume plugin. >> ?Change the ffmpeg plugin code to compile it using VS. >> ?Change the qtdemux plugin code to compile it using VS. >> ?Compile libmad for Windows and add the libraries to the Gstreamer's >> dependencies to provide mp3 decoding support. >> ?Add the mad plugin to OAbuild. >> ?Add the asfdemux plugin to OAbuild. >> ?Remove the Direct Show wrapper and use the ffmepg plugin for all the >> decoding stuff. >> ?Add the xvid plugin to OAbuild. >> ?Change the xvid plugin code to complile it usin VS. >> ?Add the mpeg2decoder plugin to OAbuild. >> ?Add the mpegstream plugin to OAbuild. >> >> To do: >> ?Make the x264 plugin work >> ?Improve mpeg2 decoding/demuxing support >> ?Add dvd navigation support >> >> >> You can find the installers at: >> http://forja.rediris.es/frs/?group_id=96 >> >> >> Andoni Morales Alastruey >> Andr?s Colubri >> >> >> ------------------------------------------------------------------------------ >> Check out the new SourceForge.net Marketplace. >> It is the best place to buy or sell services for >> just about anything Open Source. >> http://p.sf.net/sfu/Xq1LFB >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > ------------------------------------------------------------------------------ > Check out the new SourceForge.net Marketplace. > It is the best place to buy or sell services for > just about anything Open Source. > http://p.sf.net/sfu/Xq1LFB > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From msmith at xiph.org Wed Jan 7 19:34:18 2009 From: msmith at xiph.org (Michael Smith) Date: Wed, 7 Jan 2009 10:34:18 -0800 Subject: [gst-devel] FLV demuxing and decoding In-Reply-To: References: Message-ID: <3c1737210901071034w647241f5ne4ff1586b1cd3b1c@mail.gmail.com> On Wed, Jan 7, 2009 at 3:06 AM, Darren Staples wrote: > Hi again, > > As an experiment I've been trying to get flash video (.flv) files to play in > an x-window. > I can get a video-test signal to display using: "gst-launch-0.10 > videotestsrc ! Ximagesink", so I know the basic infra-structure is working > :) > > As for getting .flv files to play I've tried variations on: "gst-launch-0.10 > filesrc location=file.flv ! flv_demux ! ffdec_flv ! ximagesink". > But I keep getting an internal data stream error at the flvdemux element. The suggestions from other people to use decodebin or decodebin2 rather than manually selecting demuxers/decoders are good - you should do that. That's not what's causing the problem here though. ximagesink can only display RGB image data, and most (almost all) video decoders output some form of YUV data. So, you need to either use a sink that can directly handle YUV data (like xvimagesink - this is recommended if you can use it), or you need to convert. The 'ffmpegcolorspace' element can do this conversion. For some videos, you'll also need 'videoscale' if you're using ximagesink, because ximagesink can only display square pixels, and some videos have non-square pixels. Mike From msmith at xiph.org Wed Jan 7 19:39:20 2009 From: msmith at xiph.org (Michael Smith) Date: Wed, 7 Jan 2009 10:39:20 -0800 Subject: [gst-devel] 3gp demux plugin working on PC but not on target In-Reply-To: <89635EF2797FED4B9EF9769F80CEABC81AF3B7@EXGZ01.sasken.com> References: <89635EF2797FED4B9EF9769F80CEABC81AF3B7@EXGZ01.sasken.com> Message-ID: <3c1737210901071039r3299b2c7of849e583c86eb06e@mail.gmail.com> On Wed, Jan 7, 2009 at 4:02 AM, Rahul Verma wrote: > Hi, > > >> That said, why aren't you using qtdemux? GStreamer includes a fairly >> full-featured, high quality demuxer for >> quicktime-and-qt-derived-formats like 3gp, and it works well already. > > We use the qtdemux extensively and find it very stable. We are developing > out 3gp plugin as an experiment just to learn more about the GStreamer > framework. Fair enough. > >> Without the source to your plugin, all we can say about this >> particular crash is "it's presumably a bug in your plugin, you'll need >> to fix it". Really, this doesn't help too much. You'll need to attach a debugger and see where things are going wrong. It could be that one of the libraries gstreamer uses was compiled wrongly, so things are going wrong "randomly", or it could be a bug in your plugin - we can't guess. You might also find running it on a PC using a tool like valgrind helpful - it might show you something you're doing wrong that just luckily doesn't cause a crash on the PC, but does on your target system. Mike From bilboed at gmail.com Thu Jan 8 07:58:44 2009 From: bilboed at gmail.com (Edward Hervey) Date: Thu, 08 Jan 2009 07:58:44 +0100 Subject: [gst-devel] New GStreamer binaries for Windows based on OAbuild. In-Reply-To: <772db3280901070836x2157d230tec3c731932140972@mail.gmail.com> References: <772db3280901070342sfd0de76j9ca1169e29116972@mail.gmail.com> <180a127d0901070756w23217a3dp7a7386880b047a7b@mail.gmail.com> <772db3280901070836x2157d230tec3c731932140972@mail.gmail.com> Message-ID: <1231397924.24278.5.camel@putamadre> Hi, We're about to switch the whole of GStreamer to git soon, I think it would be nice if we could centralize OABuild alongside GStreamer in the same repositories. We already have a jhbuild module there. Tell us if you're interested, and other GStreamer hackers/devs can also raise their voice if they're against/for that idea, Edward On Wed, 2009-01-07 at 17:36 +0100, Andoni Morales wrote: > Sure! > We are planning to create a project in sourceforge to encourage every > to work on adding new plugins. > > Regards > Andoni Morales > > 2009/1/7 Julien Isorce > Hi > > Nice, could you plan to add opengl plugin in the next > release ? > > http://diracvideo.org/git?p=gst-plugins-gl.git;a=summary > > Sincerely > > Julien > > 2009/1/7 Andoni Morales > > During the last couple of months we have been working > on porting GStreamer to Windows using the OAbuild > project. We introduced some modifications to OAbuild > in order to improve compatibility and provide video > decoding and encoding in a larger number of codecs. > > Changelog: > > ?Compile ffmpeg with mingw/msys with full codec > support. > ?Remove the ffmpeg project from OAbuild and link the > ffmpeg plugin to the mingw/msys ffmpeg libraries. > ?Remove the GLib project from OAbuild and link > GStreamer to the GLib version provided > > with the GTK+ win32 installer (in this way GStreamer > can be used by GTK applications). > ?Update liboil to a newer version to compile the > volume plugin. > ?Change the ffmpeg plugin code to compile it using VS. > ?Change the qtdemux plugin code to compile it using > VS. > > ?Compile libmad for Windows and add the libraries to > the Gstreamer's dependencies to provide mp3 decoding > support. > ?Add the mad plugin to OAbuild. > ?Add the asfdemux plugin to OAbuild. > ?Remove the Direct Show wrapper and use the ffmepg > plugin for all the decoding stuff. > ?Add the xvid plugin to OAbuild. > ?Change the xvid plugin code to complile it usin VS. > ?Add the mpeg2decoder plugin to OAbuild. > ?Add the mpegstream plugin to OAbuild. > > > To do: > ?Make the x264 plugin work > ?Improve mpeg2 decoding/demuxing support > ?Add dvd navigation support > > > You can find the installers at: > http://forja.rediris.es/frs/?group_id=96 > > > Andoni Morales Alastruey > Andr?s Colubri > > > ------------------------------------------------------------------------------ > Check out the new SourceForge.net Marketplace. > It is the best place to buy or sell services for > just about anything Open Source. > http://p.sf.net/sfu/Xq1LFB > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Check out the new SourceForge.net Marketplace. > It is the best place to buy or sell services for > just about anything Open Source. > http://p.sf.net/sfu/Xq1LFB > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > Check out the new SourceForge.net Marketplace. > It is the best place to buy or sell services for > just about anything Open Source. > http://p.sf.net/sfu/Xq1LFB > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From bilboed at gmail.com Thu Jan 8 08:02:58 2009 From: bilboed at gmail.com (Edward Hervey) Date: Thu, 08 Jan 2009 08:02:58 +0100 Subject: [gst-devel] Tr : Fwd: Let people know what happens there in PALESTINE >>> In-Reply-To: <103383.79008.qm@web24404.mail.ird.yahoo.com> References: <103383.79008.qm@web24404.mail.ird.yahoo.com> Message-ID: <1231398178.24278.9.camel@putamadre> Hi, This is a DEVELOPER MAILING LIST ! We try to stay agnostic in our discussions regarding issues of politics, religions and other personal matters. We all have our personal sensibility for those matters, but it has NOTHING to do with the work done here. Please do *NOT* send this kind of mail unless it has a relationship with GStreamer. Edward On Wed, 2009-01-07 at 12:29 +0000, souilem fadi wrote: > > I know it may not be the appropriate ML , but believe me when i > received this mail from an arabic friend , i finally found that all > we heard here on TV about Isreal fighting terrorists in gaza > strip was just lies. > i do execuse , but i don't appologize for forwarding this mail > > > SOUILEM FADI > > IT engineer, Voice&video over IP developer > > > > ----- Message transf?r? ---- > De : ? SAMY MABROUK ? > Envoy? le : Lundi, 5 Janvier 2009, 14h16mn 27s > Objet : Fwd: Let people know what happens there in PALESTINE >>> > > > ---------- Forwarded message ---------- > > > > > > > > > Dear friends and colleagues; > > The following is self-explanatory and talks for itself! > > > > > > > > > > > Plzzzzz Keep forwarding this e-mail. Let people know > what happens to us in Palestine. > Por favor reenvia ese mail. El mundo debe saber que > pasa alla. > ????? ????? ?? ????? ??? ???????? ???? ?????? ???? > ???? ???? ???? ?? ?????? > > > > > > cid:4A20CAA4-489E-414A-8EBB-4C5ADC6980AE > > > cid:5265D532-E4E7-4D2E-97CC-454865C77A3B > > > > > cid:98164A6C-FD81-45E2-88B9-34D7C01F9810 > > > Making sure they get to school. > > > > > Helping Ladies across the street.. > > cid:1069CF37-5AD6-4D41-976A-F0F333AC4906 > > cid:A55FEC27-2D65-4C70-BD0B-CB1E35767E63 > > > > Providing childcare. > > cid:68A07634-7821-4DA7-B108-7919C8E07427 > > > > > Allowing them a place to rest (permanently) > > > cid:A9AF29B7-27CF-44C8-B27C-E980DE347235 > > > cid:DA139882-7151-4D23-AF97-173274D8CDB2 > > > Access to Health care. > > > > > > > Construction projects (demolition) > > > cid:C7678670-1311-4E60-A29A-2B9F10E5CE3A > > cid:57CEBA77-E3E5-4501-9FE7-7A8DA9CB3382 > > > > > Respecting American and British pacifist resisters > (such as American Rachel Corrie) > > cid:D1F22BD6-5015-43CC-BD99-52918B83DDE6 > > > cid:264BA6AB-8DD8-4118-BD7D-6293C572840F > > cid:4F9080E7-511B-44A5-B1B0-FCF529C7E325 > > > > And others. > > > cid:BB99878D-6C7E-44BF-B347-4CACB1548D65 > > > > > > > > And if you are not satisfied, now, with the truth the > following pictures are war crimes as defined by the > UN, The Hague and the Geneva Convention. > > > > > > > > Using images of your enemy dead or alive (violation) > > > cid:F08E28B4-3CA0-428D-8ECC-4683C85A3F8D > > > > Human shields (violation) > > > cid:134D38D4-8085-43C6-ABB2-1A367DE377A3 > > > > Live Burial Torture (violation) > > > cid:418B16EF-AFA6-4484-970A-F2084D28F97C > > > And as a last resort, Execution (violation) > > > > > cid:BC489237-762E-4DAB-93AC-D3EDAFD7841D > > > > These IDF soldiers have faces... I can clearly see > them...Cant you? Why are they not being prosecuted? > Because it is systematic process that is driven by the > government designed to force the people of Palestine > into exile so Israel can claim all the land and > resources. > > > This where my American tax dollars are going, do you > know where your tax dollars are at? TAKE THE TIME TO > FIND THE TRUTH. So many lives depend on it I, like so > many Americans, am Caucasian, non-Arab, and religious. > I can no longer sit back with good conscience and do > nothing while my government is supporting the types of > terrorist actions that we have condemned Muslim > Fundamentalist for. Call your Congressman and Senator, > send an email to the White House and demand that our > government negotiate FAIRLY with both sides and bring > a fair and just solution to Palestine and Israel . > > > > > cid:827F2105-4BD1-466B-9DDF-436C13BB6BDD > (CRUELTY OF ISRAEL > > > > > PLS SEND IT TO YOUR FRIENDS > > > > > > > > > > > > > > > > > > ______________________________________________________ > Life on your PC is safer, easier, and more enjoyable > with Windows Vista?. See how > > > > > > ______________________________________________________________ > Glissez - d?placez : c'est si simple de partager vos photos > sur Windows Live? Photos ! > > ______________________________________________________________ > D?couvrez tout ce que Windows Live a ? vous apporter ! > > ______________________________________________________________ > Tous vos amis discutent sur Messenger, et vous ? T?l?chargez > Messenger, c'est gratuit ! > > ______________________________________________________________ > Organisez une f?te en un tour de main : invitation, photos, > blog? > > ______________________________________________________________ > Tous vos amis discutent sur Messenger, et vous ? T?l?chargez > Messenger, c'est gratuit ! > > > > > > > --~--~---------~--~----~------------~-------~--~----~ > You received this message because you are subscribed to the Google > Groups "RT5_insat" group. > To post to this group, send email to RT5_insat at googlegroups.com > For more options, visit this group at > http://groups.google.com/group/RT5_insat > -~----------~----~----~----~------~----~------~--~--- > > > > > > -- > "Wisdom lies not in obtaining knowledge, but in using it in the right > way" - > kroesjnov > > ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ > MABROUK Samy > Security Analyst at National Agency for Computer Security > Computer Emergency Response Team CERT-TCC > > Professional mail: samy.mabrouk at ansi.tn > > > ------------------------------------------------------------------------------ > Check out the new SourceForge.net Marketplace. > It is the best place to buy or sell services for > just about anything Open Source. > http://p.sf.net/sfu/Xq1LFB > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From vaisakh.n at nestgroup.net Thu Jan 8 11:52:39 2009 From: vaisakh.n at nestgroup.net (vaisakh.n) Date: Thu, 08 Jan 2009 16:22:39 +0530 Subject: [gst-devel] gst-editor-0.10 Message-ID: <4965DAF7.6070105@nestgroup.net> while making the gst-editor-0.10 (cvs version) i got an error like this how can i solve this /***************************/ make[2]: Leaving directory `/home/Data/gst-editor-0.10/src' Making all in po make[2]: Entering directory `/home/Data/gst-editor-0.10/po' make[2]: *** No rule to make target `LINGUAS', needed by `Makefile'. Stop. make[2]: Leaving directory `/home/Data/gst-editor-0.10/po' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/home/Data/gst-editor-0.10' /*************************/ help me.. thanks in advance ***** Confidentiality Statement/Disclaimer ***** This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. From lfarkas at lfarkas.org Thu Jan 8 13:21:07 2009 From: lfarkas at lfarkas.org (Farkas Levente) Date: Thu, 08 Jan 2009 13:21:07 +0100 Subject: [gst-devel] New GStreamer binaries for Windows based on OAbuild. In-Reply-To: <772db3280901070342sfd0de76j9ca1169e29116972@mail.gmail.com> References: <772db3280901070342sfd0de76j9ca1169e29116972@mail.gmail.com> Message-ID: <4965EFB3.7030009@lfarkas.org> hi, do you have any description what you change and what you made? i'm mainly working with the mingw sig for fedora to be able to push mingw-gstreamer* into fedora. most of the basic stuff are ready and it's only time when we able to push gstreamer stuff too. but it'd help a lot if you can send patches and compile commands what you used to create these builds. thanks in advance. yours. Andoni Morales wrote: > During the last couple of months we have been working on porting > GStreamer to Windows using the OAbuild project. We introduced some > modifications to OAbuild in order to improve compatibility and provide > video decoding and encoding in a larger number of codecs. > > Changelog: > > ?Compile ffmpeg with mingw/msys with full codec support. > ?Remove the ffmpeg project from OAbuild and link the ffmpeg plugin to > the mingw/msys ffmpeg libraries. > ?Remove the GLib project from OAbuild and link GStreamer to the GLib > version provided > with the GTK+ win32 installer (in this way GStreamer can be used by GTK > applications). > ?Update liboil to a newer version to compile the volume plugin. > ?Change the ffmpeg plugin code to compile it using VS. > ?Change the qtdemux plugin code to compile it using VS. > ?Compile libmad for Windows and add the libraries to the Gstreamer's > dependencies to provide mp3 decoding support. > ?Add the mad plugin to OAbuild. > ?Add the asfdemux plugin to OAbuild. > ?Remove the Direct Show wrapper and use the ffmepg plugin for all the > decoding stuff. > ?Add the xvid plugin to OAbuild. > ?Change the xvid plugin code to complile it usin VS. > ?Add the mpeg2decoder plugin to OAbuild. > ?Add the mpegstream plugin to OAbuild. > > To do: > ?Make the x264 plugin work > ?Improve mpeg2 decoding/demuxing support > ?Add dvd navigation support > > > You can find the installers at: > http://forja.rediris.es/frs/?group_id=96 > > > Andoni Morales Alastruey > Andr?s Colubri > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------------ > Check out the new SourceForge.net Marketplace. > It is the best place to buy or sell services for > just about anything Open Source. > http://p.sf.net/sfu/Xq1LFB > > > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -- Levente "Si vis pacem para bellum!" From ylatuya at gmail.com Thu Jan 8 14:31:33 2009 From: ylatuya at gmail.com (Andoni Morales) Date: Thu, 8 Jan 2009 14:31:33 +0100 Subject: [gst-devel] New GStreamer binaries for Windows based on OAbuild. In-Reply-To: <1231397924.24278.5.camel@putamadre> References: <772db3280901070342sfd0de76j9ca1169e29116972@mail.gmail.com> <180a127d0901070756w23217a3dp7a7386880b047a7b@mail.gmail.com> <772db3280901070836x2157d230tec3c731932140972@mail.gmail.com> <1231397924.24278.5.camel@putamadre> Message-ID: <772db3280901080531r39a12d2ic2b58882a01f0106@mail.gmail.com> We're about to switch the whole of GStreamer to git soon, I think it > would be nice if we could centralize OABuild alongside GStreamer in the > same repositories. We already have a jhbuild module there. > > Tell us if you're interested, and other GStreamer hackers/devs can > also raise their voice if they're against/for that idea, > I think it's a good idea to centralize OABuild along GStreamer in the same repository. The problem is that the OABuild "method" requires changes in the code of some plugins. A posible solution is to use preprocesor directives as these changes just affect to 3 or 4 plugins. Besides this methods requires external depencies (GLib, xvid, x264, oil, etc..), that cannot be complied with Visual Studio and must me complied with mingw/msys. This is not a big problem as they can be added in the OAbuild module. Regards, Andoni Morales -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilboed at gmail.com Thu Jan 8 15:01:50 2009 From: bilboed at gmail.com (Edward Hervey) Date: Thu, 08 Jan 2009 15:01:50 +0100 Subject: [gst-devel] New GStreamer binaries for Windows based on OAbuild. In-Reply-To: <772db3280901080531r39a12d2ic2b58882a01f0106@mail.gmail.com> References: <772db3280901070342sfd0de76j9ca1169e29116972@mail.gmail.com> <180a127d0901070756w23217a3dp7a7386880b047a7b@mail.gmail.com> <772db3280901070836x2157d230tec3c731932140972@mail.gmail.com> <1231397924.24278.5.camel@putamadre> <772db3280901080531r39a12d2ic2b58882a01f0106@mail.gmail.com> Message-ID: <1231423310.5684.0.camel@localhost> On Thu, 2009-01-08 at 14:31 +0100, Andoni Morales wrote: > We're about to switch the whole of GStreamer to git soon, I think it > would be nice if we could centralize OABuild alongside > GStreamer in the > same repositories. We already have a jhbuild module there. > > Tell us if you're interested, and other GStreamer > hackers/devs can > also raise their voice if they're against/for that idea, > > I think it's a good idea to centralize OABuild along GStreamer in the > same repository. The problem is that the OABuild "method" requires > changes in the code of some plugins. A posible solution is to use > preprocesor directives as these changes just affect to 3 or 4 > plugins. Can those changes be pushed upstream ? > Besides this methods requires external depencies (GLib, xvid, x264, > oil, etc..), that cannot be complied with Visual Studio and must me > complied with mingw/msys. This is not a big problem as they can be > added in the OAbuild module. > > Regards, > Andoni Morales > ------------------------------------------------------------------------------ > Check out the new SourceForge.net Marketplace. > It is the best place to buy or sell services for > just about anything Open Source. > http://p.sf.net/sfu/Xq1LFB > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From dstaples at friendmts.co.uk Thu Jan 8 15:13:29 2009 From: dstaples at friendmts.co.uk (Darren Staples) Date: Thu, 08 Jan 2009 14:13:29 +0000 Subject: [gst-devel] Issue with decodebin ? Message-ID: Hi again :) Is there an issue with using the decodebin element programmatically ? This command line works fine: gst-launch filesrc location=file.flv ! decodebin ! ffmpegcolorspace ! ximagesink But when I try to code it up as follows: (some code left out for clarity) // Create gstreamer elements pipeline = gst_pipeline_new( "video" ); source = gst_element_factory_make( "filesrc" , "src" ); decoder = gst_element_factory_make( "decodebin" , "decoder" ); colorspace1 = gst_element_factory_make( "ffmpegcolorspace" , "colorspace1" ); sink = gst_element_factory_make( "ximagesink" , "sink" ); if ( !pipeline || !source || !decoder || !colorspace1 || !sink ) { g_printerr( "One element could not be created. Exiting.\n" ); return -1; } // Set the input filename to the source element g_object_set( G_OBJECT( source ) , "location" , "file.flv" , NULL ); // Add all elements into the pipeline gst_bin_add_many( GST_BIN( pipeline ) , source , decoder , colorspace1 , sink , NULL ); // Start linking the elements together gst_element_link_many( source , decoder , colorspace1 , sink , NULL ); // Set the pipeline to "playing" state gst_element_set_state( pipeline , GST_STATE_PLAYING ); I get an Ointernal data flow error? as you would see when linking failed. Any ideas why this might be please ? Does the uridecodebin work any better ? Thanks for all the help so far :D Darren *** FRIEND MTS LIMITED EMAIL CONFIDENTIALITY *** This email message and any attachments may contain information which is confidential or privileged and is intended for the sole use of the person to whom it is addressed. If you are not the intended recipient, be aware that any disclosures, copying, distribution or use of the contents is prohibited. If you have received this email message in error, please notify our office by telephone (+44 (0)121 633 2000) or email (admin at friendmts.co.uk) immediately. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: From alessandro.d at gmail.com Thu Jan 8 15:21:21 2009 From: alessandro.d at gmail.com (Alessandro Decina) Date: Thu, 8 Jan 2009 15:21:21 +0100 Subject: [gst-devel] New GStreamer binaries for Windows based on OAbuild. In-Reply-To: <1231423310.5684.0.camel@localhost> References: <772db3280901070342sfd0de76j9ca1169e29116972@mail.gmail.com> <180a127d0901070756w23217a3dp7a7386880b047a7b@mail.gmail.com> <772db3280901070836x2157d230tec3c731932140972@mail.gmail.com> <1231397924.24278.5.camel@putamadre> <772db3280901080531r39a12d2ic2b58882a01f0106@mail.gmail.com> <1231423310.5684.0.camel@localhost> Message-ID: <37eb22810901080621n38a472ecma2508e2f1920ef03@mail.gmail.com> On Thu, Jan 8, 2009 at 3:01 PM, Edward Hervey wrote: > On Thu, 2009-01-08 at 14:31 +0100, Andoni Morales wrote: >> I think it's a good idea to centralize OABuild along GStreamer in the >> same repository. The problem is that the OABuild "method" requires >> changes in the code of some plugins. A posible solution is to use >> preprocesor directives as these changes just affect to 3 or 4 >> plugins. > > Can those changes be pushed upstream ? For ffmpeg for example, no. They are syntax changes needed to make msvc happy but that compile just fine with gcc. And they are A LOT (thousands of lines). Ciao Alessandro From wim.taymans at gmail.com Thu Jan 8 15:51:06 2009 From: wim.taymans at gmail.com (Wim Taymans) Date: Thu, 8 Jan 2009 15:51:06 +0100 Subject: [gst-devel] Issue with decodebin ? In-Reply-To: References: Message-ID: decodebin dynamically creates source pads as can be seen with gst-inspect. Please re-read Capter 8 of the application development manual. Wim On Thu, Jan 8, 2009 at 3:13 PM, Darren Staples wrote: > Hi again :) > > Is there an issue with using the decodebin element programmatically ? > > This command line works fine: > gst-launch filesrc location=file.flv ! decodebin ! ffmpegcolorspace ! > ximagesink > > But when I try to code it up as follows: (some code left out for clarity) > > // Create gstreamer elements > pipeline = gst_pipeline_new( "video" ); > source = gst_element_factory_make( "filesrc" , "src" ); > decoder = gst_element_factory_make( "decodebin" , "decoder" ); > colorspace1 = gst_element_factory_make( "ffmpegcolorspace" , "colorspace1" > ); > sink = gst_element_factory_make( "ximagesink" , "sink" ); > > if ( !pipeline || !source || !decoder || !colorspace1 || !sink ) > { > g_printerr( "One element could not be created. Exiting.\n" ); > return -1; > } > > // Set the input filename to the source element > g_object_set( G_OBJECT( source ) , "location" , "file.flv" , NULL ); > > // Add all elements into the pipeline > gst_bin_add_many( GST_BIN( pipeline ) , source , decoder , colorspace1 , > sink , NULL ); > > // Start linking the elements together > gst_element_link_many( source , decoder , colorspace1 , sink , NULL ); > > // Set the pipeline to "playing" state > gst_element_set_state( pipeline , GST_STATE_PLAYING ); > > I get an 'internal data flow error' as you would see when linking failed. > > Any ideas why this might be please ? Does the uridecodebin work any better ? > > Thanks for all the help so far :D > > > Darren > > > > *** FRIEND MTS LIMITED EMAIL CONFIDENTIALITY *** > > This email message and any attachments may contain information which is > confidential or privileged and is intended for the sole use of the person to > whom it is addressed. If you are not the intended recipient, be aware that > any disclosures, copying, distribution or use of the contents is > prohibited. If you have received this email message in error, please notify > our office by telephone (+44 (0)121 633 2000) or email > (admin at friendmts.co.uk) immediately. Thank you. > > ------------------------------------------------------------------------------ > Check out the new SourceForge.net Marketplace. > It is the best place to buy or sell services for > just about anything Open Source. > http://p.sf.net/sfu/Xq1LFB > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From ylatuya at gmail.com Thu Jan 8 16:15:33 2009 From: ylatuya at gmail.com (Andoni Morales) Date: Thu, 8 Jan 2009 16:15:33 +0100 Subject: [gst-devel] New GStreamer binaries for Windows based on OAbuild. In-Reply-To: <37eb22810901080621n38a472ecma2508e2f1920ef03@mail.gmail.com> References: <772db3280901070342sfd0de76j9ca1169e29116972@mail.gmail.com> <180a127d0901070756w23217a3dp7a7386880b047a7b@mail.gmail.com> <772db3280901070836x2157d230tec3c731932140972@mail.gmail.com> <1231397924.24278.5.camel@putamadre> <772db3280901080531r39a12d2ic2b58882a01f0106@mail.gmail.com> <1231423310.5684.0.camel@localhost> <37eb22810901080621n38a472ecma2508e2f1920ef03@mail.gmail.com> Message-ID: <772db3280901080715k76e63954g58ac7c87ded5bedd@mail.gmail.com> > > > For ffmpeg for example, no. They are syntax changes needed to make > msvc happy but that compile just fine with gcc. And they are A LOT > (thousands of lines). > I don't use MSVC to build ffmpeg. All the external libraries are built using mingw/msys and then I add to the OAbuild "builddeps" dir the includes, dll's and impot libraries used by OABuild. The problem with the original OABuild method is that all the external libraries are built using MSVC, and that's why it doesn't provide support for many external plugins, because not all the external dependencies can be compiled using MSVC. The changes in the GStreamer code are just a few lines. Take a look at http://codeanticode.wordpress.com/2008/09/12/the-gstreamer-adventure-part-i-creating-a-windows-installer/. For the ffmpeg plugin the lines to changes are 4: - In gstffmpegenc.c do: replace in_plugin = first_avcodec; by in_plugin = av_codec_next(NULL); and in_plugin = in_plugin->next; by in_plugin = av_codec_next(in_plugin); - In gstffmpegdec.c, do: replace in_plugin = first_avcodec; by in_plugin = av_codec_next(NULL); and in_plugin = in_plugin->next; by in_plugin = av_codec_next(in_plugin); So these changes can be pusshed upstream -------------- next part -------------- An HTML attachment was scrubbed... URL: From ole.andre.ravnas at tandberg.com Thu Jan 8 16:27:19 2009 From: ole.andre.ravnas at tandberg.com (=?iso-8859-1?Q?Ole_Andre_Vadla_Ravn=E5s?=) Date: Thu, 8 Jan 2009 16:27:19 +0100 Subject: [gst-devel] New GStreamer binaries for Windows based on OAbuild. In-Reply-To: <772db3280901080531r39a12d2ic2b58882a01f0106@mail.gmail.com> References: <772db3280901070342sfd0de76j9ca1169e29116972@mail.gmail.com><180a127d0901070756w23217a3dp7a7386880b047a7b@mail.gmail.com><772db3280901070836x2157d230tec3c731932140972@mail.gmail.com><1231397924.24278.5.camel@putamadre> <772db3280901080531r39a12d2ic2b58882a01f0106@mail.gmail.com> Message-ID: Fra: Andoni Morales [mailto:ylatuya at gmail.com] Sendt: Thursday, January 08, 2009 2:32 PM Til: Discussion of the development of GStreamer Emne: Re: [gst-devel] New GStreamer binaries for Windows based on OAbuild. We're about to switch the whole of GStreamer to git soon, I think it would be nice if we could centralize OABuild alongside GStreamer in the same repositories. We already have a jhbuild module there. Tell us if you're interested, and other GStreamer hackers/devs can also raise their voice if they're against/for that idea, I think it's a good idea to centralize OABuild along GStreamer in the same repository. The problem is that the OABuild "method" requires changes in the code of some plugins. A posible solution is to use preprocesor directives as these changes just affect to 3 or 4 plugins. Besides this methods requires external depencies (GLib, xvid, x264, oil, etc..), that cannot be complied with Visual Studio and must me complied with mingw/msys. This is not a big problem as they can be added in the OAbuild module. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ylatuya at gmail.com Thu Jan 8 16:31:17 2009 From: ylatuya at gmail.com (Andoni Morales) Date: Thu, 8 Jan 2009 16:31:17 +0100 Subject: [gst-devel] New GStreamer binaries for Windows based on OAbuild. In-Reply-To: <772db3280901080715k76e63954g58ac7c87ded5bedd@mail.gmail.com> References: <772db3280901070342sfd0de76j9ca1169e29116972@mail.gmail.com> <180a127d0901070756w23217a3dp7a7386880b047a7b@mail.gmail.com> <772db3280901070836x2157d230tec3c731932140972@mail.gmail.com> <1231397924.24278.5.camel@putamadre> <772db3280901080531r39a12d2ic2b58882a01f0106@mail.gmail.com> <1231423310.5684.0.camel@localhost> <37eb22810901080621n38a472ecma2508e2f1920ef03@mail.gmail.com> <772db3280901080715k76e63954g58ac7c87ded5bedd@mail.gmail.com> Message-ID: <772db3280901080731g2b5fd21cm19175b29a0b27744@mail.gmail.com> By the way, It would be nice to update in the web page the windows binaries with these ones. Regards, Andoni Morales -------------- next part -------------- An HTML attachment was scrubbed... URL: From oleavr at gmail.com Thu Jan 8 17:14:24 2009 From: oleavr at gmail.com (=?ISO-8859-1?Q?Ole_Andr=E9_Vadla_Ravn=E5s?=) Date: Thu, 8 Jan 2009 17:14:24 +0100 Subject: [gst-devel] New GStreamer binaries for Windows based on OAbuild. In-Reply-To: <772db3280901080531r39a12d2ic2b58882a01f0106@mail.gmail.com> References: <772db3280901070342sfd0de76j9ca1169e29116972@mail.gmail.com> <180a127d0901070756w23217a3dp7a7386880b047a7b@mail.gmail.com> <772db3280901070836x2157d230tec3c731932140972@mail.gmail.com> <1231397924.24278.5.camel@putamadre> <772db3280901080531r39a12d2ic2b58882a01f0106@mail.gmail.com> Message-ID: (Sorry for spamming the list with an empty reply before sending this one, I was fighting with a brain-dead e-mail client to get the quoting right and ended up hitting the wrong hotkey.) On Thu, Jan 8, 2009 at 2:31 PM, Andoni Morales wrote: > We're about to switch the whole of GStreamer to git soon, I think it >> >> would be nice if we could centralize OABuild alongside GStreamer in the >> same repositories. We already have a jhbuild module there. >> >> Tell us if you're interested, and other GStreamer hackers/devs can >> also raise their voice if they're against/for that idea, > > I think it's a good idea to centralize OABuild along GStreamer in the same > repository. The problem is that the OABuild "method" requires changes in the > code of some plugins. A posible solution is to use preprocesor directives > as these changes just affect to 3 or 4 plugins. > Besides this methods requires external depencies (GLib, xvid, x264, oil, > etc..), that cannot be complied with Visual Studio and must me complied with > mingw/msys. This is not a big problem as they can be added in the OAbuild > module. I would vote against pushing the project files upstream as OABuild's basic design is broken (see below). As of recently, pending announcement today actually :), OABuild is officially dead and no longer maintained by me. Its replacement is OAHBuild, which is OABuild completely redone from scratch by Haakon Sporsheim. Some of its advantages: - Isn't monolithic like OABuild, so the different modules are independent in the sense that you no longer need them placed in a certain layout in the filesystem. In other words you can choose to install GLib and GStreamer -core from MSIs, and build /only/ -bad yourself if you're fixing bugs in a plugin, you don't need to build any of the dependencies. This also means that the project files are completely local to the module and can thus be pushed upstream. The only dependency if you want to build things yourself is the common OAH component. - Integrates with, and makes heavy use of, pkg-config. - Integrated easy way of building .msm packages. - Designed to work with the Express Edition of MSVS 2008, the commercial version is no longer a requirement to build it using the IDE. - Cleaner, less project files. OAHBuild can be found here: https://code.launchpad.net/oah The existing GStreamer modules in OABuild aren't yet forward-ported to OAH, but this is work in progress. Cheers, Ole Andr? From oleavr at gmail.com Thu Jan 8 17:23:00 2009 From: oleavr at gmail.com (=?ISO-8859-1?Q?Ole_Andr=E9_Vadla_Ravn=E5s?=) Date: Thu, 8 Jan 2009 17:23:00 +0100 Subject: [gst-devel] New GStreamer binaries for Windows based on OAbuild. In-Reply-To: References: <772db3280901070342sfd0de76j9ca1169e29116972@mail.gmail.com> <180a127d0901070756w23217a3dp7a7386880b047a7b@mail.gmail.com> <772db3280901070836x2157d230tec3c731932140972@mail.gmail.com> <1231397924.24278.5.camel@putamadre> <772db3280901080531r39a12d2ic2b58882a01f0106@mail.gmail.com> Message-ID: On Thu, Jan 8, 2009 at 5:14 PM, Ole Andr? Vadla Ravn?s wrote: > (Sorry for spamming the list with an empty reply before sending this one, > I was fighting with a brain-dead e-mail client to get the quoting right and > ended up hitting the wrong hotkey.) > > On Thu, Jan 8, 2009 at 2:31 PM, Andoni Morales wrote: >> We're about to switch the whole of GStreamer to git soon, I think it >>> >>> would be nice if we could centralize OABuild alongside GStreamer in the >>> same repositories. We already have a jhbuild module there. >>> >>> Tell us if you're interested, and other GStreamer hackers/devs can >>> also raise their voice if they're against/for that idea, >> >> I think it's a good idea to centralize OABuild along GStreamer in the same >> repository. The problem is that the OABuild "method" requires changes in the >> code of some plugins. A posible solution is to use preprocesor directives >> as these changes just affect to 3 or 4 plugins. >> Besides this methods requires external depencies (GLib, xvid, x264, oil, >> etc..), that cannot be complied with Visual Studio and must me complied with >> mingw/msys. This is not a big problem as they can be added in the OAbuild >> module. (...) > OAHBuild can be found here: > https://code.launchpad.net/oah Oops, not quite the right URL, should've been: https://launchpad.net/oah Cheers, Ole Andr? From ylatuya at gmail.com Thu Jan 8 18:35:52 2009 From: ylatuya at gmail.com (Andoni Morales) Date: Thu, 8 Jan 2009 18:35:52 +0100 Subject: [gst-devel] New GStreamer binaries for Windows based on OAbuild. In-Reply-To: References: <772db3280901070342sfd0de76j9ca1169e29116972@mail.gmail.com> <180a127d0901070756w23217a3dp7a7386880b047a7b@mail.gmail.com> <772db3280901070836x2157d230tec3c731932140972@mail.gmail.com> <1231397924.24278.5.camel@putamadre> <772db3280901080531r39a12d2ic2b58882a01f0106@mail.gmail.com> Message-ID: <772db3280901080935x2b0800b3la3fd3b7a80de696d@mail.gmail.com> > > > Its replacement is OAHBuild, which is OABuild completely redone from > scratch > by Haakon Sporsheim. Good to know this! I'll be waiting until it's ready to port all these changes to OAH. -------------- next part -------------- An HTML attachment was scrubbed... URL: From MFonnesbeck at sorenson.com Thu Jan 8 19:39:22 2009 From: MFonnesbeck at sorenson.com (Merrick Fonnesbeck) Date: Thu, 8 Jan 2009 11:39:22 -0700 Subject: [gst-devel] Passing video/audio stream to another RTP library Message-ID: I have a pipeline that currently streams video using GStreamer to another location forming a simple internet call application. gst-launch v4l2src ! video/x-raw-yuv,width=176,height=144,framerate=8/1 ! hantro4200enc ! rtph263pay ! udpsink host= port= I want to use SIP to coordinate session information with the connection to the destination and I have a SIP library framework that I already own a license for that I would like to use (yes I know that the N810 come with Sofia-SIP), and this SIP library also comes with RTP capabilities. I am wondering if it is possible for GStreamer to pass it's information off to this other library of code and let it take care of the RTP transport of streaming video data to the destination location and receive incoming data and pass it along into GStreamer's own elements for processing and displaying on the screen? If anyone knows or has any ideas, please let me know. Thanks. Merrick -------------- next part -------------- An HTML attachment was scrubbed... URL: From olivier.crete at collabora.co.uk Thu Jan 8 19:56:34 2009 From: olivier.crete at collabora.co.uk (Olivier =?ISO-8859-1?Q?Cr=EAte?=) Date: Thu, 08 Jan 2009 13:56:34 -0500 Subject: [gst-devel] Passing video/audio stream to another RTP library In-Reply-To: References: Message-ID: <1231440995.7153.16.camel@TesterTop3.tester.ca> On Thu, 2009-01-08 at 11:39 -0700, Merrick Fonnesbeck wrote: > I have a pipeline that currently streams video using GStreamer to > another location forming a simple internet call application. > > gst-launch v4l2src ! > video/x-raw-yuv,width=176,height=144,framerate=8/1 ! hantro4200enc ! > rtph263pay ! udpsink host= port= > > I want to use SIP to coordinate session information with the > connection to the destination and I have a SIP library framework that > I already own a license for that I would like to use (yes I know that > the N810 come with Sofia-SIP), and this SIP library also comes with > RTP capabilities. I am wondering if it is possible for GStreamer to > pass it's information off to this other library of code and let it > take care of the RTP transport of streaming video data to the > destination location and receive incoming data and pass it along into > GStreamer's own elements for processing and displaying on the screen? > If anyone knows or has any ideas, please let me know. Thanks. Use appsrc and appsink. If you are on a N810, you probably want to backport the version that have just been merged into gst-plugins-base. -- Olivier Cr?te olivier.crete at collabora.co.uk Collabora Ltd -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: This is a digitally signed message part URL: From ob.lutz at gmail.com Thu Jan 8 22:48:04 2009 From: ob.lutz at gmail.com (OB Lutz) Date: Thu, 8 Jan 2009 16:48:04 -0500 Subject: [gst-devel] Grabbing snapshots out of live stream Message-ID: Hey all I'm trying to write a pipeline that will take video in from a network source, do some transcoding, and send it out to a network sink. No problem there. But I need to be able to get preview snapshots of the stream as its running, either on demand, or just have it continuously dump out frames to a known file periodically. I'm doing some experiments using a v4lsrc and xvimagesink as my fake network source and sink for now. I've found the 'tee' block that looks promising to split my stream into 2, so one can go on its way to the network sink, and the other could maybe write to a file, but all my experiments have only shown me a single frame in my xvimagesink. Am I going down the right path trying to use 'tee' or is there some other set of blocks I should use? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From thx1138 at earthlink.net Fri Jan 9 04:23:10 2009 From: thx1138 at earthlink.net (thx1138) Date: Thu, 08 Jan 2009 19:23:10 -0800 Subject: [gst-devel] VST Audio Plug in approach using Gstreamer and Linux OS with ARM Cortex A8 processors In-Reply-To: <49635BB0.5080002@hora-obscura.de> Message-ID: On 1/6/09 5:25 AM, "Stefan Kost" wrote: > thx1138 schrieb: >> Hello, >> >> We are presently using VST as a plug-in for audio applications and the >> question came up today is it possible for VST plug-ins to work with >> G-Streamer? Our VST's are migrating from PC/Mac Intel host and desire to use >> VST approach with Gstreamer for ARM Cortex A8 processor. >> >> We have Linux Gstreamer as our Linux interface and we are wondering what has >> been done in this area already to look at the API interfaces etc. > > If those are your own vst plugins I would port them to gstreamer. For vst > effects you could use GstAudioFilter as a baseclass > (gst-plugins-base/gst-libs/gst/audio/). > > If you want to keep the plugins as they are, you would need to write a brige > plugin. It would load the vst plugins as a host an externaly appear as a > gstreamer plugin. > > Stefan > >> >> Regards, >> >> Terry >> >> >> ----------------------------------------------------------------------------->> - >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel Hello, I have discovered that VST Plugins can be used with WINE interface to Linux. This was done using an Intel and AMD processor. Does anyone have some good links to doing the same using ARM 11 or Cortex A8 processors? We are suing Gstreamer for our plug-in interface for Multichannel matrix audio decoding and post processing. I am new to this as all of my earlier work was DSP based and now I must migrate to ARM all of my Dolby, DTS, AAC, MP3 etc. decoding and post processing. Any suggestions would be great. Terry From sastraxi at gmail.com Fri Jan 9 05:20:24 2009 From: sastraxi at gmail.com (Cameron Gorrie) Date: Thu, 8 Jan 2009 23:20:24 -0500 Subject: [gst-devel] VST Audio Plug in approach using Gstreamer and Linux OS with ARM Cortex A8 processors In-Reply-To: References: <49635BB0.5080002@hora-obscura.de> Message-ID: <448b46400901082020o1fc05050r6d61d6baf86797aa@mail.gmail.com> You might take a look at the "JUCE" project; I think there's something in there to do with cross-platform (OSX, Linux, Windows) VST-type plugins. http://www.rawmaterialsoftware.com/juce/ Cam 2009/1/8 thx1138 : > On 1/6/09 5:25 AM, "Stefan Kost" wrote: > >> thx1138 schrieb: >>> Hello, >>> >>> We are presently using VST as a plug-in for audio applications and the >>> question came up today is it possible for VST plug-ins to work with >>> G-Streamer? Our VST's are migrating from PC/Mac Intel host and desire to use >>> VST approach with Gstreamer for ARM Cortex A8 processor. >>> >>> We have Linux Gstreamer as our Linux interface and we are wondering what has >>> been done in this area already to look at the API interfaces etc. >> >> If those are your own vst plugins I would port them to gstreamer. For vst >> effects you could use GstAudioFilter as a baseclass >> (gst-plugins-base/gst-libs/gst/audio/). >> >> If you want to keep the plugins as they are, you would need to write a brige >> plugin. It would load the vst plugins as a host an externaly appear as a >> gstreamer plugin. >> >> Stefan >> >>> >>> Regards, >>> >>> Terry >>> >>> >>> > ----------------------------------------------------------------------------->> > - >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> ------------------------------------------------------------------------------ >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > Hello, > > I have discovered that VST Plugins can be used with WINE interface to Linux. > This was done using an Intel and AMD processor. Does anyone have some good > links to doing the same using ARM 11 or Cortex A8 processors? We are suing > Gstreamer for our plug-in interface for Multichannel matrix audio decoding > and post processing. > > I am new to this as all of my earlier work was DSP based and now I must > migrate to ARM all of my Dolby, DTS, AAC, MP3 etc. decoding and post > processing. > > Any suggestions would be great. > > Terry > > > ------------------------------------------------------------------------------ > Check out the new SourceForge.net Marketplace. > It is the best place to buy or sell services for > just about anything Open Source. > http://p.sf.net/sfu/Xq1LFB > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From haakon.sporsheim at gmail.com Fri Jan 9 09:24:20 2009 From: haakon.sporsheim at gmail.com (Haakon Sporsheim) Date: Fri, 9 Jan 2009 09:24:20 +0100 Subject: [gst-devel] New GStreamer binaries for Windows based on OAbuild. In-Reply-To: References: <772db3280901070342sfd0de76j9ca1169e29116972@mail.gmail.com> <180a127d0901070756w23217a3dp7a7386880b047a7b@mail.gmail.com> <772db3280901070836x2157d230tec3c731932140972@mail.gmail.com> <1231397924.24278.5.camel@putamadre> <772db3280901080531r39a12d2ic2b58882a01f0106@mail.gmail.com> Message-ID: <27bc9d90901090024w20bb2aedvf81368895a9d675b@mail.gmail.com> > Its replacement is OAHBuild, which is OABuild completely redone from > scratch > by Haakon Sporsheim. > > Some of its advantages: > - Isn't monolithic like OABuild, so the different modules are independent > in the > sense that you no longer need them placed in a certain layout in the > filesystem. > In other words you can choose to install GLib and GStreamer -core from > MSIs, > and build /only/ -bad yourself if you're fixing bugs in a plugin, > you don't need to > build any of the dependencies. This also means that the project files are > completely local to the module and can thus be pushed upstream. The only > dependency if you want to build things yourself is the common OAH > component. > - Integrates with, and makes heavy use of, pkg-config. > - Integrated easy way of building .msm packages. > - Designed to work with the Express Edition of MSVS 2008, the commercial > version is no longer a requirement to build it using the IDE. > - Cleaner, less project files. > I would also like to point out (once again) that OAH uses pkg-config for compiler and linker switches instead of project dependencies/references in Visual Studio. Hence, there is no monolitic "master" build/solution file, but separate build/solution file for each module which could reside in each respective module repository. For GStreamer this would be (to follow the OAH convention) in win32/oah, which is beside the existing win32/vs*. The existing GStreamer modules in OABuild aren't yet forward-ported to OAH, > but > this is work in progress. > Yes, I've already started the work, but currently I'm sort of waiting for the upcoming cvs->git migration. Ref. Edward Hervey's mail to mailinglist a couple of days ago. :) Regards, Haakon -------------- next part -------------- An HTML attachment was scrubbed... URL: From peter.kjellerstedt at axis.com Fri Jan 9 11:19:37 2009 From: peter.kjellerstedt at axis.com (Peter Kjellerstedt) Date: Fri, 9 Jan 2009 11:19:37 +0100 Subject: [gst-devel] G726 RTP payload packetizer In-Reply-To: <011f01c970bf$041db510$04000100@telxsi.com> References: <011f01c970bf$041db510$04000100@telxsi.com> Message-ID: Our audio guy here (who wrote the G.726 payloader) has looked at it and it seems correct that it does not follow RFC 3551 correctly for 24/40 kbps. This is due to historical reasons. :( Could you add an entry to the bugzilla database for this, and he will take care of it? //Peter > -----Original Message----- > From: Kumar [mailto:kumarkm at tataelxsi.co.in] > Sent: den 7 januari 2009 12:57 > To: 'Kumar'; Gstreamer-Devel (E-mail) > Cc: dev-gstreamer > Subject: Re: [gst-devel] G726 RTP payload packetizer > > Hi all, > > To add one more point, is this implementation covers the > recommendations of RFC 3551.? > > Thanks, > Kumar KM > > -----Original Message----- > From: Kumar [mailto:kumarkm at tataelxsi.co.in] > Sent: Wednesday, January 07, 2009 5:12 PM > To: Gstreamer-Devel (E-mail) > Cc: 'dev-gstreamer at axis.com' > Subject: G726 RTP payload packetizer > > Hi all, > > I have been using the G726 RTP packetizer for streaming from > Gstreamer to VLC player. when I use a Little -endian(LE) input > data, for 32kbps & 16kbps it works fine, without any noise. But > for 24kbps & 40 kbps it gives noise for LE data. When I use all > test streams with BE, then it works fine. Does this G726 RTP > packetizer accept data in Big-endian(BE)? Can somebody give > more info on this. > > Thanks in advance! > > Regards, > Kumar KM From kumarkm at tataelxsi.co.in Fri Jan 9 11:33:47 2009 From: kumarkm at tataelxsi.co.in (Kumar) Date: Fri, 9 Jan 2009 16:03:47 +0530 Subject: [gst-devel] G726 RTP payload packetizer In-Reply-To: Message-ID: <000201c97245$c1b256d0$04000100@telxsi.com> Hi Peter, Thanks for the informaion. I will add an entry to the bugzilla database on this issue. Regards, Kumar -----Original Message----- From: Peter Kjellerstedt [mailto:peter.kjellerstedt at axis.com] Sent: Friday, January 09, 2009 3:50 PM To: kumarkm at tataelxsi.co.in; Discussion of the development of GStreamer Cc: dev-gstreamer Subject: RE: [gst-devel] G726 RTP payload packetizer Our audio guy here (who wrote the G.726 payloader) has looked at it and it seems correct that it does not follow RFC 3551 correctly for 24/40 kbps. This is due to historical reasons. :( Could you add an entry to the bugzilla database for this, and he will take care of it? //Peter > -----Original Message----- > From: Kumar [mailto:kumarkm at tataelxsi.co.in] > Sent: den 7 januari 2009 12:57 > To: 'Kumar'; Gstreamer-Devel (E-mail) > Cc: dev-gstreamer > Subject: Re: [gst-devel] G726 RTP payload packetizer > > Hi all, > > To add one more point, is this implementation covers the > recommendations of RFC 3551.? > > Thanks, > Kumar KM > > -----Original Message----- > From: Kumar [mailto:kumarkm at tataelxsi.co.in] > Sent: Wednesday, January 07, 2009 5:12 PM > To: Gstreamer-Devel (E-mail) > Cc: 'dev-gstreamer at axis.com' > Subject: G726 RTP payload packetizer > > Hi all, > > I have been using the G726 RTP packetizer for streaming from > Gstreamer to VLC player. when I use a Little -endian(LE) input > data, for 32kbps & 16kbps it works fine, without any noise. But > for 24kbps & 40 kbps it gives noise for LE data. When I use all > test streams with BE, then it works fine. Does this G726 RTP > packetizer accept data in Big-endian(BE)? Can somebody give > more info on this. > > Thanks in advance! > > Regards, > Kumar KM The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. -------------- next part -------------- A non-text attachment was scrubbed... Name: winmail.dat Type: application/ms-tnef Size: 2400 bytes Desc: not available URL: From ylatuya at gmail.com Fri Jan 9 16:20:33 2009 From: ylatuya at gmail.com (Andoni Morales) Date: Fri, 9 Jan 2009 16:20:33 +0100 Subject: [gst-devel] New GStreamer binaries for Windows based on OAbuild. In-Reply-To: <27bc9d90901090024w20bb2aedvf81368895a9d675b@mail.gmail.com> References: <772db3280901070342sfd0de76j9ca1169e29116972@mail.gmail.com> <180a127d0901070756w23217a3dp7a7386880b047a7b@mail.gmail.com> <772db3280901070836x2157d230tec3c731932140972@mail.gmail.com> <1231397924.24278.5.camel@putamadre> <772db3280901080531r39a12d2ic2b58882a01f0106@mail.gmail.com> <27bc9d90901090024w20bb2aedvf81368895a9d675b@mail.gmail.com> Message-ID: <772db3280901090720k34b4b2cfvb34fc645b74c13fa@mail.gmail.com> > > I would also like to point out (once again) that OAH uses pkg-config for > compiler and linker switches instead of project dependencies/references in > Visual Studio. Hence, there is no monolitic "master" build/solution file, > but separate build/solution file for each module which could reside in each > respective module repository. For GStreamer this would be (to follow the OAH > convention) in win32/oah, which is beside the existing win32/vs*. I think the adoption of pkg-config for compiler and linker switches is an exelent choice as it will make easier to compile plugins with external depencies, which was one the problems of OABuild. -------------- next part -------------- An HTML attachment was scrubbed... URL: From jack.foster11 at gmail.com Fri Jan 9 16:39:00 2009 From: jack.foster11 at gmail.com (Jack Foster) Date: Fri, 9 Jan 2009 10:39:00 -0500 Subject: [gst-devel] Finding Out Negotiated Caps? Message-ID: Hi gang! First, is there an easy way to find out the Caps that are coming out of a udpsink block? I'd like to know so I can statically attach them as the input for the udpsrc block in the receiver. Secondly, I realise that any dynamic negotiation of codecs happens outside of rtp but how would I get started if I wanted to do that? Cheers, Jack From mapping at sheerman-chase.org.uk Fri Jan 9 18:17:39 2009 From: mapping at sheerman-chase.org.uk (mapping at sheerman-chase.org.uk) Date: Fri, 09 Jan 2009 17:17:39 +0000 Subject: [gst-devel] getting GdkPixbuf from gdkpixbufsink Message-ID: <20090109171739.z9n9gkeqwo80ogoo@maggie.enixns.com> Hello all, I am looking for some help getting the object GdkPixbuf from gdkpixbufsink. I am working in image processing so I want to get the image buffer for each frame. My priorities are to have a reliable and conceptually simple way of getting the buffer. I have started by modifying gst-player-0.0.0, changed it to use the gdkpixbufsink sink and waiting to process messages on the bus. I am seeing the element/pixbuf messages without a problem. I then call the following on the bus message: buff = gst_structure_get_value (msg->structure,"pixbuf"); which gives me a non-zero pointer. I was expecting that is a GdkPixbuf object. The problem I get is when I do GDK_IS_PIXBUF(buff) it returns false. Any ideas why? Or am I meant to retrieve "last-pixbuf" from the sink directly? If more detail is needed, the code is here: http://www.sheerman-chase.org.uk/temp/gst-player-pixel.tar.gz Tim From t.i.m at zen.co.uk Fri Jan 9 19:56:39 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Fri, 09 Jan 2009 18:56:39 +0000 Subject: [gst-devel] getting GdkPixbuf from gdkpixbufsink In-Reply-To: <20090109171739.z9n9gkeqwo80ogoo@maggie.enixns.com> References: <20090109171739.z9n9gkeqwo80ogoo@maggie.enixns.com> Message-ID: <1231527400.1319.18.camel@mini.centricular.net> On Fri, 2009-01-09 at 17:17 +0000, mapping at sheerman-chase.org.uk wrote: Hi, > I have started by modifying gst-player-0.0.0, changed it to use the > gdkpixbufsink sink and waiting to process messages on the bus. I am > seeing the element/pixbuf messages without a problem. I then call the > following on the bus message: > > buff = gst_structure_get_value (msg->structure,"pixbuf"); > > which gives me a non-zero pointer. I was expecting that is a GdkPixbuf > object. The problem I get is when I do GDK_IS_PIXBUF(buff) it returns > false. Any ideas why? It returns a const GValue * from which you can retrieve the GdkPixbuf object via pixbuf = g_value_dup_object (val); (which gives you a reference to the pixbuf which you need to drop with g_object_unref(pixbuf) when you don't need it any longer) > Or am I meant to retrieve "last-pixbuf" from the sink directly? You can do that as well. The "last-pixbuf" property is useful if you want to check the pixbuf in your own time/interval and you're just interested in the current pixbuf and not interested whether you're missing any or not. If you want to evaluate each and every pixbuf, you need to process the messages. There's also an example at: http://webcvs.freedesktop.org/gstreamer/gst-plugins-good/tests/icles/gdkpixbufsink-test.c?view=markup Cheers -Tim From gstelzz at yahoo.fr Fri Jan 9 20:18:37 2009 From: gstelzz at yahoo.fr (Aurelien Grimaud) Date: Fri, 09 Jan 2009 20:18:37 +0100 Subject: [gst-devel] Finding Out Negotiated Caps? In-Reply-To: References: Message-ID: <4967A30D.5030106@yahoo.fr> Jack Foster a ?crit : > Hi gang! > > First, is there an easy way to find out the Caps that are coming out > of a udpsink block? I'd like to know so I can statically attach them > as the input for the udpsrc block in the receiver. > > Secondly, I realise that any dynamic negotiation of codecs happens > outside of rtp but how would I get started if I wanted to do that? > You need a negotiation protocol for your codecs. Take a look at RTSP for audio / video streaming. A rtsp client is included in gstreamer (gst-inspect rtspsrc). There is also a rtsp server written with gstreamer : http://blogs.gnome.org/uraeus/2008/10/13/gstreamer-rtsp-server/ For telephone communications, SIP would be use. You then need a sip stack for signaling protocol, gstreamer will be handling media. > Cheers, > Jack > > ------------------------------------------------------------------------------ > Check out the new SourceForge.net Marketplace. > It is the best place to buy or sell services for > just about anything Open Source. > http://p.sf.net/sfu/Xq1LFB > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From zeenix at gmail.com Sat Jan 10 00:36:44 2009 From: zeenix at gmail.com (Zeeshan Ali (Khattak)) Date: Sat, 10 Jan 2009 01:36:44 +0200 Subject: [gst-devel] Is there any UPnP Media sever that can share real audio on RTSP? In-Reply-To: <610EAA62A0FD314D9761697EEC820CBE64A076@banasiexb01.ASIA.ROOT.PRI> References: <610EAA62A0FD314D9761697EEC820CBE64A076@banasiexb01.ASIA.ROOT.PRI> Message-ID: <38294b740901091536t40a75848t20abc45c78935d1c@mail.gmail.com> Hi! 2009/1/5 Nagamani Veerappa : > Hi, > > Is there any Gstreamer application/Plugin or anything else, that can work > like UPnP Media server sharing real audio on RTSP? No! infact i don't see how that would work. You could instead make a plugin for Rygel[1] that export media from any RTSP source. The DVB plugin of Rygel already exports the RTSP URIs from DVB Daemon[2] but there is nothing stopping a plugin from implementing a complete RTSP server on it's own. If you go for that option, I suggest you use gst-rtsp-server[3] (like DVB Daemon does) to easily implement an RTSP server. -- Regards, Zeeshan Ali (Khattak) FSF member#5124 [1] http://live.gnome.org/Rygel [2] http://live.gnome.org/DVBDaemon [3] http://git.collabora.co.uk/?p=gst-rtsp-server.git;a=summary From thaytan at noraisin.net Sat Jan 10 00:41:43 2009 From: thaytan at noraisin.net (Jan Schmidt) Date: Fri, 09 Jan 2009 23:41:43 +0000 Subject: [gst-devel] 2nd pre-releases of Core/Base/Bad/Python available In-Reply-To: <1231286471.6539.17.camel@fancy-ubuntu> References: <1231279326.6539.5.camel@fancy-ubuntu> <1231286471.6539.17.camel@fancy-ubuntu> Message-ID: <1231544503.21410.4.camel@fancy-ubuntu> Hi all, A new set of pre-releases of all 4 modules are now available at: http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.21.3.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.21.3.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-bad/pre/gst-plugins-bad-0.10.9.3.tar.bz2 and http://gstreamer.freedesktop.org/src/gst-python/pre/gst-python-0.10.13.3.tar.bz2 These fix a variety of small bugs and omissions. Please give them some testing love. File bugs at http://bugzilla.gnome.org If necessary, there'll be new pre-releases on Tuesday (13 Jan). The final release tarballs are expected on Monday week, the 19th January. After that, we'll be staying frozen for a bit while we switch all the repositories over to git. Cheers, Jan. On Wed, 2009-01-07 at 00:01 +0000, Jan Schmidt wrote: > Hi all, > > The first pre-releases of the 4 modules are now available at: > http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.21.2.tar.bz2 > http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.21.2.tar.bz2 > http://gstreamer.freedesktop.org/src/gst-plugins-bad/pre/gst-plugins-bad-0.10.9.2.tar.bz2 > and > http://gstreamer.freedesktop.org/src/gst-python/pre/gst-python-0.10.13.2.tar.bz2 > > Please test, and file any bugs at http://bugzilla.gnome.org/ ! > > Expect new pre-releases on Friday if needed. Check the release plan to > find out about the release schedule: > http://www.gstreamer.net/wiki/ReleasePlanning2009-1 > > Cheers, > Jan. > > > On Tue, 2009-01-06 at 22:02 +0000, Jan Schmidt wrote: > > Hi all, > > > > All 4 of the Core, Base Plugins, Bad Plugins and Python bindings modules > > are now frozen for releases. The Bad Plugins aren't normally in this > > cycle. They're there this month to handle appsrc and appsink moving from > > -bad to -base at last, and audioresample trading places with the much > > better speexresample. Speexresample is now in -base, but registers with > > the name 'audioresample'. The old audioresample is now in -bad as > > 'legacyresample' > > > > See the release plan for the first few months of this year > > (http://www.gstreamer.net/wiki/ReleasePlanning2009-1) for more detail, > > and expect the pre-releases later tonight. > > > > Cheers, > > Jan. -- Jan Schmidt From liangzhihong1984 at 126.com Sat Jan 10 03:51:43 2009 From: liangzhihong1984 at 126.com (liangzhihong1984) Date: Sat, 10 Jan 2009 10:51:43 +0800 (CST) Subject: [gst-devel] About how to get the running parameters of GStreamer application Message-ID: <28621127.394161231555903191.JavaMail.coremail@bj126app15.126.com> Hi, wim.I really appreciate your help. And I have another question, how can I get the sessionID of my current RTP Session? Best Regards, Leon Liang -------------- next part -------------- An HTML attachment was scrubbed... URL: From 1985renzf at gmail.com Sat Jan 10 13:00:14 2009 From: 1985renzf at gmail.com (zhenfeng ren) Date: Sat, 10 Jan 2009 20:00:14 +0800 Subject: [gst-devel] how "gst-inspect" work Message-ID: hi,everyone: I cross compile the Gstreamer for ARM. The problem is that the "gst-inspect" could not find any plugins which are under GST_PLUGIN_PATH. When I run gdb, I find that the "main()" of "gst-inspect" is in "gst-run.c". I try set a breakpoint at "main()" in "gst-inspect.c", but I failed. I cannot find out how "gst-inspect" should work. Could anyone tell me how "gst-inspect" find the plugins ? That is , if I compile a plugin and copy is to GST_PLUGIN_PATH, how could "gst-inspect" know that there is a new plugin? -- Thanks, Zhenfeng Ren From ensonic at hora-obscura.de Sat Jan 10 16:58:54 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sat, 10 Jan 2009 17:58:54 +0200 Subject: [gst-devel] VST Audio Plug in approach using Gstreamer and Linux OS with ARM Cortex A8 processors In-Reply-To: References: Message-ID: <4968C5BE.6090303@hora-obscura.de> hi, thx1138 schrieb: > On 1/6/09 5:25 AM, "Stefan Kost" wrote: > >> thx1138 schrieb: >>> Hello, >>> >>> We are presently using VST as a plug-in for audio applications and the >>> question came up today is it possible for VST plug-ins to work with >>> G-Streamer? Our VST's are migrating from PC/Mac Intel host and desire to use >>> VST approach with Gstreamer for ARM Cortex A8 processor. >>> >>> We have Linux Gstreamer as our Linux interface and we are wondering what has >>> been done in this area already to look at the API interfaces etc. >> If those are your own vst plugins I would port them to gstreamer. For vst >> effects you could use GstAudioFilter as a baseclass >> (gst-plugins-base/gst-libs/gst/audio/). >> >> If you want to keep the plugins as they are, you would need to write a brige >> plugin. It would load the vst plugins as a host an externaly appear as a >> gstreamer plugin. >> >> Stefan >> >>> Regards, >>> >>> Terry >>> >>> >>> > ----------------------------------------------------------------------------->> > - >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> ------------------------------------------------------------------------------ >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > Hello, > > I have discovered that VST Plugins can be used with WINE interface to Linux. > This was done using an Intel and AMD processor. Does anyone have some good > links to doing the same using ARM 11 or Cortex A8 processors? We are suing > Gstreamer for our plug-in interface for Multichannel matrix audio decoding > and post processing. the wine approach wont help you. The idea behind that is to reuse the plugins compiles on windows. It emulates the windows api as far as it needed and runs the dlls. I do this for buzzmachine plugins for gstreamer, pitfdll does this for windows codecs. If you do this for vst, you could use windows vft plugins under linux, but only on x86 (and only on 32 bit for 32bit plugins). If you want to run them on arm, then you need to recompile them. Using qemu wouldn't give you the needed perormance. Stefan > > I am new to this as all of my earlier work was DSP based and now I must > migrate to ARM all of my Dolby, DTS, AAC, MP3 etc. decoding and post > processing. > > Any suggestions would be great. > > Terry > > > ------------------------------------------------------------------------------ > Check out the new SourceForge.net Marketplace. > It is the best place to buy or sell services for > just about anything Open Source. > http://p.sf.net/sfu/Xq1LFB > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From thx1138 at earthlink.net Sat Jan 10 19:10:46 2009 From: thx1138 at earthlink.net (thx1138) Date: Sat, 10 Jan 2009 10:10:46 -0800 Subject: [gst-devel] VST Audio Plug in approach using Gstreamer and Linux OS with ARM Cortex A8 processors In-Reply-To: <4968C5BE.6090303@hora-obscura.de> Message-ID: On 1/10/09 7:58 AM, "Stefan Kost" wrote: > hi, > > thx1138 schrieb: >> On 1/6/09 5:25 AM, "Stefan Kost" wrote: >> >>> thx1138 schrieb: >>>> Hello, >>>> >>>> We are presently using VST as a plug-in for audio applications and the >>>> question came up today is it possible for VST plug-ins to work with >>>> G-Streamer? Our VST's are migrating from PC/Mac Intel host and desire to >>>> use >>>> VST approach with Gstreamer for ARM Cortex A8 processor. >>>> >>>> We have Linux Gstreamer as our Linux interface and we are wondering what >>>> has >>>> been done in this area already to look at the API interfaces etc. >>> If those are your own vst plugins I would port them to gstreamer. For vst >>> effects you could use GstAudioFilter as a baseclass >>> (gst-plugins-base/gst-libs/gst/audio/). >>> >>> If you want to keep the plugins as they are, you would need to write a brige >>> plugin. It would load the vst plugins as a host an externaly appear as a >>> gstreamer plugin. >>> >>> Stefan >>> >>>> Regards, >>>> >>>> Terry >>>> >>>> >>>> >> ----------------------------------------------------------------------------- >> >> >> - >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> gstreamer-devel at lists.sourceforge.net >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> ---------------------------------------------------------------------------- >>> -- >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> Hello, >> >> I have discovered that VST Plugins can be used with WINE interface to Linux. >> This was done using an Intel and AMD processor. Does anyone have some good >> links to doing the same using ARM 11 or Cortex A8 processors? We are suing >> Gstreamer for our plug-in interface for Multichannel matrix audio decoding >> and post processing. > > the wine approach wont help you. The idea behind that is to reuse the plugins > compiles on windows. It emulates the windows api as far as it needed and runs > the dlls. I do this for buzzmachine plugins for gstreamer, pitfdll does this > for > windows codecs. If you do this for vst, you could use windows vft plugins > under > linux, but only on x86 (and only on 32 bit for 32bit plugins). > > If you want to run them on arm, then you need to recompile them. Using qemu > wouldn't give you the needed perormance. > > Stefan > >> >> I am new to this as all of my earlier work was DSP based and now I must >> migrate to ARM all of my Dolby, DTS, AAC, MP3 etc. decoding and post >> processing. >> >> Any suggestions would be great. >> >> Terry >> >> >> ----------------------------------------------------------------------------->> - >> Check out the new SourceForge.net Marketplace. >> It is the best place to buy or sell services for >> just about anything Open Source. >> http://p.sf.net/sfu/Xq1LFB >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > Check out the new SourceForge.net Marketplace. > It is the best place to buy or sell services for > just about anything Open Source. > http://p.sf.net/sfu/Xq1LFB > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel Hello Herr Kost, So, If I understand your response, VST plugins compiled for Windowx/MacOSX would need to be recompiled to run as plug-ins using Gstreamer. Wine just give a Windows emulation for x86 emulation I gather. Our software manager has decided to build our Dolby, DTS, etc. plug-ins to use Gstreamer as interface which I believe is fine, I just need other VST applications to be ported to Gstreamer. I wish I had some quick way to evaluate Gstreamer plug-ins versus VST approach. Perhaps this will take some effort to gather benchmarks on latency. Danke fur alles From ylatuya at gmail.com Sun Jan 11 00:17:16 2009 From: ylatuya at gmail.com (Andoni Morales) Date: Sun, 11 Jan 2009 00:17:16 +0100 Subject: [gst-devel] Video capturer Message-ID: <772db3280901101517o23841b0cm9747ed9a410f9aba@mail.gmail.com> Hi, I'm trying to implement a video capturer using the following pipeline: source->decodebin->tee-->displaybin '->encodebin The pipeline is started without linking the encodebin to the tee. When I click the rec button, the tee is linked to the encodebin which encodes the video stream into a file. The problem is that the output file displays at the begining x seconds of the first frame. For example: Time: 0:00:00 -> pipeline changes to PLAYING state. Time: 0:00:10 -> rec button clicked, encodebin added to the pipeline and linked to the tee. Time: 0:00:30 -> stop button clicked, sending eos event to the pipeline. The output file displays the first frame for 10 seconds, and then 20 seconds of the captured stream. Do I have to fire a new segment event before linking the encodebin to the tee? Regards, Andoni Morales -------------- next part -------------- An HTML attachment was scrubbed... URL: From liangzhihong1984 at 126.com Sun Jan 11 04:09:00 2009 From: liangzhihong1984 at 126.com (liangzhihong1984) Date: Sun, 11 Jan 2009 11:09:00 +0800 (CST) Subject: [gst-devel] About how to use gstreamer signals Message-ID: <27469524.34431231643340096.JavaMail.coremail@bj126app88.126.com> Hi, everyone. I would like to use gstreamer singals to get some running parameters, but I don't know how to use them in my project. For example: I got a signal named "get-stats" in multiudpsink, and it is defined like this: GValueArray*user_function (GstMultiUDPSink *gstmultiudpsink, gchar *host, gint port, gpointer user_data); Get the statistics of the client with destination host and port. gstmultiudpsink :the sink on which the signal is emitted host :the hostname/IP address of the client to get stats on port :the port of the client to get stats on user_data :user data set when the signal handler was connected. Returns :a GValueArray of uint64: bytes_sent, packets_sent, connect_time (in epoch seconds), disconnect_time (in epoch seconds) My question is when I use this signal with g_signal_emit_by_name, how to return the GValueArray Object, where is it stored? Or any other better ways? Can anybody give me some help. Thank you in advance. Best Regards, Leon Liang -------------- next part -------------- An HTML attachment was scrubbed... URL: From liangzhihong1984 at 126.com Sun Jan 11 06:27:05 2009 From: liangzhihong1984 at 126.com (liangzhihong1984) Date: Sun, 11 Jan 2009 13:27:05 +0800 (CST) Subject: [gst-devel] Error with "get-internal-session" on GstRtpBin Message-ID: <11147386.56171231651625651.JavaMail.coremail@bj126app15.126.com> I defined a GstRtpBin and the initialize it like: GstElement *rtpbin; rtpbin = gst_element_factory_make("gstrtpbin", "rtpbin"); then I sent an "get-internal-session" signal to rtpbin with session ID to get the session: g_signal_emit_by_name(G_OBJECT(m_gstrtpbin), "get-internal-session", 0, rSession); but, error occured: signal name `get-internal-session' is invalid for instance `0x87ad188' I 've no idea about this error, because the "get-internal-session" signal is a signal of the GstRtpBin object. Please give me some points. Thank you. Best Regards, Leon Liang -------------- next part -------------- An HTML attachment was scrubbed... URL: From ensonic at hora-obscura.de Sun Jan 11 10:31:06 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sun, 11 Jan 2009 11:31:06 +0200 Subject: [gst-devel] VST Audio Plug in approach using Gstreamer and Linux OS with ARM Cortex A8 processors In-Reply-To: References: Message-ID: <4969BC5A.8040406@hora-obscura.de> thx1138 schrieb: > On 1/10/09 7:58 AM, "Stefan Kost" wrote: > >> hi, >> >> thx1138 schrieb: >>> On 1/6/09 5:25 AM, "Stefan Kost" wrote: >>> >>>> thx1138 schrieb: >>>>> Hello, >>>>> >>>>> We are presently using VST as a plug-in for audio applications and the >>>>> question came up today is it possible for VST plug-ins to work with >>>>> G-Streamer? Our VST's are migrating from PC/Mac Intel host and desire to >>>>> use >>>>> VST approach with Gstreamer for ARM Cortex A8 processor. >>>>> >>>>> We have Linux Gstreamer as our Linux interface and we are wondering what >>>>> has >>>>> been done in this area already to look at the API interfaces etc. >>>> If those are your own vst plugins I would port them to gstreamer. For vst >>>> effects you could use GstAudioFilter as a baseclass >>>> (gst-plugins-base/gst-libs/gst/audio/). >>>> >>>> If you want to keep the plugins as they are, you would need to write a brige >>>> plugin. It would load the vst plugins as a host an externaly appear as a >>>> gstreamer plugin. >>>> >>>> Stefan >>>> >>>>> Regards, >>>>> >>>>> Terry >>>>> >>>>> >>>>> >>> ----------------------------------------------------------------------------- >>> - >>>>> _______________________________________________ >>>>> gstreamer-devel mailing list >>>>> gstreamer-devel at lists.sourceforge.net >>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> ---------------------------------------------------------------------------- >>>> -- >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> gstreamer-devel at lists.sourceforge.net >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> Hello, >>> >>> I have discovered that VST Plugins can be used with WINE interface to Linux. >>> This was done using an Intel and AMD processor. Does anyone have some good >>> links to doing the same using ARM 11 or Cortex A8 processors? We are suing >>> Gstreamer for our plug-in interface for Multichannel matrix audio decoding >>> and post processing. >> the wine approach wont help you. The idea behind that is to reuse the plugins >> compiles on windows. It emulates the windows api as far as it needed and runs >> the dlls. I do this for buzzmachine plugins for gstreamer, pitfdll does this >> for >> windows codecs. If you do this for vst, you could use windows vft plugins >> under >> linux, but only on x86 (and only on 32 bit for 32bit plugins). >> >> If you want to run them on arm, then you need to recompile them. Using qemu >> wouldn't give you the needed perormance. >> >> Stefan >> >>> I am new to this as all of my earlier work was DSP based and now I must >>> migrate to ARM all of my Dolby, DTS, AAC, MP3 etc. decoding and post >>> processing. >>> >>> Any suggestions would be great. >>> >>> Terry >>> >>> >>> > ----------------------------------------------------------------------------->> > - >>> Check out the new SourceForge.net Marketplace. >>> It is the best place to buy or sell services for >>> just about anything Open Source. >>> http://p.sf.net/sfu/Xq1LFB >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> ------------------------------------------------------------------------------ >> Check out the new SourceForge.net Marketplace. >> It is the best place to buy or sell services for >> just about anything Open Source. >> http://p.sf.net/sfu/Xq1LFB >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > Hello Herr Kost, > > So, If I understand your response, VST plugins compiled for Windowx/MacOSX > would need to be recompiled to run as plug-ins using Gstreamer. > > Wine just give a Windows emulation for x86 emulation I gather. > > Our software manager has decided to build our Dolby, DTS, etc. plug-ins to > use Gstreamer as interface which I believe is fine, I just need other VST > applications to be ported to Gstreamer. > > I wish I had some quick way to evaluate Gstreamer plug-ins versus VST > approach. Perhaps this will take some effort to gather benchmarks on > latency. > > Danke fur alles > The latency is more dependend on what audio i/o plugins you use and how they are configured. On windows vst is typically used with ASIO. On linux a good couterpart would be jack, but that might be too heavy for an embedded system (mixing is float based). A well configured alsa might be enough. It really depends what you want to do. Functionality wise the GStreamer plugin api is quite rich. I belive the plugins can be ported. On thig that is not covered by gstreamer is plugin uis. Thats left to the plugin host. Stefan From lazyt at gmx.net Sun Jan 11 20:11:32 2009 From: lazyt at gmx.net (Thomas Loewe) Date: Sun, 11 Jan 2009 11:11:32 -0800 (PST) Subject: [gst-devel] [neonhttpsrc] Authentication? Message-ID: <21402814.post@talk.nabble.com> Hi, is it possible to use neonhttpsrc with Authentication? I've tried http://user:pass at server without Success. Regards, Thomas -- View this message in context: http://www.nabble.com/-neonhttpsrc--Authentication--tp21402814p21402814.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From vaisakh.n at nestgroup.net Mon Jan 12 06:09:47 2009 From: vaisakh.n at nestgroup.net (vaisakh.n) Date: Mon, 12 Jan 2009 10:39:47 +0530 Subject: [gst-devel] gst-inspect Message-ID: <496AD09B.8010401@nestgroup.net> what is actually gst-inspect-0.10 doing.. will it hook the plugins or just prints the information ?? help me.. thanks ***** Confidentiality Statement/Disclaimer ***** This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. From msmith at xiph.org Mon Jan 12 07:30:53 2009 From: msmith at xiph.org (Michael Smith) Date: Sun, 11 Jan 2009 22:30:53 -0800 Subject: [gst-devel] gst-inspect In-Reply-To: <496AD09B.8010401@nestgroup.net> References: <496AD09B.8010401@nestgroup.net> Message-ID: <3c1737210901112230xaffa6fepd3869c694c3a2721@mail.gmail.com> On Sun, Jan 11, 2009 at 9:09 PM, vaisakh.n wrote: > what is actually gst-inspect-0.10 doing.. > will it hook the plugins or just prints the information ?? > help me.. > thanks gst-inspect-0.10 initialises gstreamer (which causes the registry to be loaded, and refreshed if needed). It then looks up information in the registry, and if you select a specific plugin, loads that plugin to get additional information about it. Mike > ***** Confidentiality Statement/Disclaimer ***** > > This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. > The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. Please ensure you don't send nonsense disclaimers like this to mailing lists. I note for the record that I'm ignoring everything this says. Mike From dhanraj.b at allaboutif.com Mon Jan 12 10:28:00 2009 From: dhanraj.b at allaboutif.com (Dhanraj Birader) Date: Mon, 12 Jan 2009 14:58:00 +0530 Subject: [gst-devel] gstreamer Message-ID: <9536d8fe0901120128k7a7092f0x7e9096402da80904@mail.gmail.com> Hi, udpsrc plugin is working perfectly on x86 but it is not working on davinci board. It is streaming correctly but video is not displaying. Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: From dhanraj.b at allaboutif.com Mon Jan 12 10:00:30 2009 From: dhanraj.b at allaboutif.com (Dhanraj Birader) Date: Mon, 12 Jan 2009 14:30:30 +0530 Subject: [gst-devel] gstreamer Message-ID: <9536d8fe0901120100n1347cb68y3e9307e0fa2d102a@mail.gmail.com> Hi, Why udpsrc is not working on davinci board. Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: From dstaples at friendmts.co.uk Mon Jan 12 11:14:15 2009 From: dstaples at friendmts.co.uk (Darren Staples) Date: Mon, 12 Jan 2009 10:14:15 +0000 Subject: [gst-devel] Multiple ximagesinks Message-ID: Is it possible to have more than one ximagesink in a pipeline ? Suppose I wanted to preview both an original of a movie file, and a monochrome/filtered version. I realise that I need a Otee? element to split the video stream into two, but I?m struggling to get to grips with the gst-launch syntax required when using a tee. My simple pipeline is: filesrc location=file.flv ! decodebin ! ffmpegcolorspace ! video/x-raw-gray ! ffmepgcolorspace ! ximagesink How would that look with a tee and a second ximagesink displaying the original colour movie ? Thanks for all your help so far ! Darren *** FRIEND MTS LIMITED EMAIL CONFIDENTIALITY *** This email message and any attachments may contain information which is confidential or privileged and is intended for the sole use of the person to whom it is addressed. If you are not the intended recipient, be aware that any disclosures, copying, distribution or use of the contents is prohibited. If you have received this email message in error, please notify our office by telephone (+44 (0)121 633 2000) or email (admin at friendmts.co.uk) immediately. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: From felipe.contreras at gmail.com Mon Jan 12 11:26:55 2009 From: felipe.contreras at gmail.com (Felipe Contreras) Date: Mon, 12 Jan 2009 12:26:55 +0200 Subject: [gst-devel] Multiple ximagesinks In-Reply-To: References: Message-ID: <94a0d4530901120226x3e574f99s97f7f8f62c6ea99c@mail.gmail.com> On Mon, Jan 12, 2009 at 12:14 PM, Darren Staples wrote: > Is it possible to have more than one ximagesink in a pipeline ? > > Suppose I wanted to preview both an original of a movie file, and a > monochrome/filtered version. I realise that I need a 'tee' element to split > the video stream into two, but I'm struggling to get to grips with the > gst-launch syntax required when using a tee. > > My simple pipeline is: > filesrc location=file.flv ! decodebin ! ffmpegcolorspace ! video/x-raw-gray > ! ffmepgcolorspace ! ximagesink > > How would that look with a tee and a second ximagesink displaying the > original colour movie ? > > Thanks for all your help so far ! This works for me: gst-launch videotestsrc ! tee name=t t. ! queue ! ximagesink t. ! queue ! ximagesink -- Felipe Contreras From julien at moutte.net Mon Jan 12 11:20:09 2009 From: julien at moutte.net (Julien Moutte) Date: Mon, 12 Jan 2009 11:20:09 +0100 Subject: [gst-devel] Multiple ximagesinks In-Reply-To: References: Message-ID: <496B1959.8080101@moutte.net> Yes, you can put as many ximagesink you want in your pipeline. Just try this : gst-launch-0.10 videotestsrc ! tee name=t ! queue ! ximagesink t. ! queue ! ximagesink Cheers, Julien Moutte Darren Staples wrote: > Is it possible to have more than one ximagesink in a pipeline ? > > Suppose I wanted to preview both an original of a movie file, and a > monochrome/filtered version. I realise that I need a ?tee? element to > split the video stream into two, but I?m struggling to get to grips with > the gst-launch syntax required when using a tee. > > My simple pipeline is: > filesrc location=file.flv ! decodebin ! ffmpegcolorspace ! > video/x-raw-gray ! ffmepgcolorspace ! ximagesink > > How would that look with a tee and a second ximagesink displaying the > original colour movie ? > > Thanks for all your help so far ! > > Darren > > > > *** FRIEND MTS LIMITED EMAIL CONFIDENTIALITY *** > > This email message and any attachments may contain information which is > confidential or privileged and is intended for the sole use of the > person to whom it is addressed. If you are not the intended recipient, > be aware that any disclosures, copying, distribution or use of the > contents is prohibited. If you have received this email message in > error, please notify our office by telephone (+44 (0)121 633 2000) or > email (admin at friendmts.co.uk ) > immediately. Thank you. > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------------ > Check out the new SourceForge.net Marketplace. > It is the best place to buy or sell services for > just about anything Open Source. > http://p.sf.net/sfu/Xq1LFB > > > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From dstaples at friendmts.co.uk Mon Jan 12 15:14:12 2009 From: dstaples at friendmts.co.uk (Darren Staples) Date: Mon, 12 Jan 2009 14:14:12 +0000 Subject: [gst-devel] Appsink questions Message-ID: 1. Can anyone point me at a non-out-of-date example for using the appsink element please ? 2. If not, what events/signals do I need to use to be able to access the raw-frame data for my movie ? 3. Should I just set the pipeline to play, or do I need to step one frame at a time (I anticipate that processing of the raw frame data will be slow enough that my application will not be able to keep up with the full frame rate) ? Thanks ! Darren *** FRIEND MTS LIMITED EMAIL CONFIDENTIALITY *** This email message and any attachments may contain information which is confidential or privileged and is intended for the sole use of the person to whom it is addressed. If you are not the intended recipient, be aware that any disclosures, copying, distribution or use of the contents is prohibited. If you have received this email message in error, please notify our office by telephone (+44 (0)121 633 2000) or email (admin at friendmts.co.uk) immediately. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: From wim.taymans at gmail.com Mon Jan 12 16:02:04 2009 From: wim.taymans at gmail.com (Wim Taymans) Date: Mon, 12 Jan 2009 16:02:04 +0100 Subject: [gst-devel] Appsink questions In-Reply-To: References: Message-ID: On Mon, Jan 12, 2009 at 3:14 PM, Darren Staples wrote: > Can anyone point me at a non-out-of-date example for using the appsink > element please ? I don't think there is out-of-data documentation for appsink. > If not, what events/signals do I need to use to be able to access the > raw-frame data for my movie ? did you check the documentation (in the source code) here: http://webcvs.freedesktop.org/gstreamer/gst-plugins-base/gst-libs/gst/app/gstappsink.c?revision=1.19&view=markup Pleas let us know what's missing. > Should I just set the pipeline to play, or do I need to step one frame at a yes, configure and link appsink as the sink, then play the pipeline and start pulling buffers. Wim > time (I anticipate that processing of the raw frame data will be slow enough > that my application will not be able to keep up with the full frame rate) ? > > Thanks ! > > Darren > > > > *** FRIEND MTS LIMITED EMAIL CONFIDENTIALITY *** > > This email message and any attachments may contain information which is > confidential or privileged and is intended for the sole use of the person to > whom it is addressed. If you are not the intended recipient, be aware that > any disclosures, copying, distribution or use of the contents is > prohibited. If you have received this email message in error, please notify > our office by telephone (+44 (0)121 633 2000) or email > (admin at friendmts.co.uk) immediately. Thank you. > > ------------------------------------------------------------------------------ > Check out the new SourceForge.net Marketplace. > It is the best place to buy or sell services for > just about anything Open Source. > http://p.sf.net/sfu/Xq1LFB > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From arnout at mind.be Mon Jan 12 16:33:15 2009 From: arnout at mind.be (Arnout Vandecappelle) Date: Mon, 12 Jan 2009 16:33:15 +0100 Subject: [gst-devel] Grabbing snapshots out of live stream In-Reply-To: References: Message-ID: <200901121633.15781.arnout@mind.be> On Thursday 08 January 2009 22:48:04 OB Lutz wrote: > I'm trying to write a pipeline that will take video in from a network > source, do some transcoding, and send it out to a network sink. No problem > there. But I need to be able to get preview snapshots of the stream as its > running, either on demand, or just have it continuously dump out frames to > a known file periodically. I'm doing some experiments using a v4lsrc and > xvimagesink as my fake network source and sink for now. I've found the > 'tee' block that looks promising to split my stream into 2, so one can go > on its way to the network sink, and the other could maybe write to a file, > but all my experiments have only shown me a single frame in my xvimagesink. > Am I going down the right path trying to use 'tee' or is there some other > set of blocks I should use? You probably need to insert a queue behind the tee. Otherwise you can get deadlocks because all sinks try to synchronize from the same thread. Failing that, can you post the gst-launch command you're using? Regards, Arnout -- Arnout Vandecappelle arnout at mind be Senior Embedded Software Architect +32-16-286540 Essensium/Mind http://www.mind.be G.Geenslaan 9, 3001 Leuven, Belgium BE 872 984 063 RPR Leuven LinkedIn profile: http://www.linkedin.com/in/arnoutvandecappelle GPG fingerprint: D206 D44B 5155 DF98 550D 3F2A 2213 88AA A1C7 C933 From mapping at sheerman-chase.org.uk Mon Jan 12 17:50:19 2009 From: mapping at sheerman-chase.org.uk (mapping at sheerman-chase.org.uk) Date: Mon, 12 Jan 2009 16:50:19 +0000 Subject: [gst-devel] getting an accurate frame number from pipeline Message-ID: <20090112165019.t9cn6h9c00wkwgsc@maggie.enixns.com> Hi, Thanks again people for help with gdkpixbufsink. I am now trying to get an accurate video frame number from a pipeline. I can change the format I am requesting but only GST_FORMAT_TIME and GST_FORMAT_PERCENT work as expected. When I ask for GST_FORMAT_BUFFERS, the query returns with the same result as GST_FORMAT_PERCENT. int video_get_pos(GstFormat fmt, const char *text) { GstQuery *query; gboolean res; query = gst_query_new_position (fmt); res = gst_element_query (pipeline, query); if (res) { gint64 pos; gst_query_parse_position (query, &fmt, &pos); printf ("position %s \t= %lld\n", text, (pos)); } else { printf ("position query failed...\n"); } gst_query_unref (query); return 0; } And I am querying with different formats like this: video_get_pos(GST_FORMAT_DEFAULT,"default"); video_get_pos(GST_FORMAT_TIME,"time "); video_get_pos(GST_FORMAT_BUFFERS,"buffers"); video_get_pos(GST_FORMAT_PERCENT,"percent"); I get a result along these lines: position default = 11040000000 position time = 10882214000 position buffers = 11040000000 position percent = 11040000000 Is this a good approach to get an accurate frame number? Or do I have to work around and get the time in nanoseconds and multiply it by the frame rate? I also have problems seeking using frame numbers. Is gstreamer trying to tell me something? full code: http://www.sheerman-chase.org.uk/temp/gst-player-pixel2009-01-12d.tar.gz Thanks, Tim From jack.foster11 at gmail.com Mon Jan 12 20:57:29 2009 From: jack.foster11 at gmail.com (Jack Foster) Date: Mon, 12 Jan 2009 14:57:29 -0500 Subject: [gst-devel] Grabbing RTSP Stats? Message-ID: Hiya Gang, I currently have an rtp/rtsp server -> client stream via gstrtpbin. Is there a way I can get access to the statistics of the sender and receiver reports? I'd like the server to know if the client missed packets or if the communication path is fully broken. Would this be the on-timeout signal? Jack From ob.lutz at gmail.com Mon Jan 12 21:24:20 2009 From: ob.lutz at gmail.com (OB Lutz) Date: Mon, 12 Jan 2009 15:24:20 -0500 Subject: [gst-devel] Grabbing snapshots out of live stream In-Reply-To: <200901121633.15781.arnout@mind.be> References: <200901121633.15781.arnout@mind.be> Message-ID: I've managed to get passed the whole single-frame issue, but am getting pipeline errors when actually running it using something closer to my final pipeline. I've reduced my problem to the following command. This is taking in raw YUY2 data(eventually to be replaced by some sort of mjpeg stream, but I have access to YUY2 currently), shoving it in a tee and writing out frames to a file. gst-launch -vvv gstrtpbin name=rtpbin \ tcpserversrc host=127.0.0.1 port=1235 ! videoparse format=YUY2 width=404 height=424 framerate=20/1 bpp=16 ! \ tee name=splitter splitter. ! queue ! typefind ! pngenc snapshot=false ! multifilesink sync=false preroll-queue-len=20 location=/tmp/output.png -vvv splitter. This starts up fine, with: Setting pipeline to PAUSED ... Pipeline is PREROLLING ... but when I start sending data to it, I error out with: /pipeline0/videoparse0.src: caps = video/x-raw-yuv, width=(int)404, height=(int)424, format=(fourcc)YUY2, framerate=(fraction)20/1, pixel_aspect_ratio=(fraction)1/1 /pipeline0/splitter.sink: caps = video/x-raw-yuv, width=(int)404, height=(int)424, format=(fourcc)YUY2, framerate=(fraction)20/1, pixel_aspect_ratio=(fraction)1/1 /pipeline0/splitter.src0: caps = video/x-raw-yuv, width=(int)404, height=(int)424, format=(fourcc)YUY2, framerate=(fraction)20/1, pixel_aspect_ratio=(fraction)1/1 /pipeline0/splitter.sink: caps = video/x-raw-yuv, width=(int)404, height=(int)424, format=(fourcc)YUY2, framerate=(fraction)20/1, pixel_aspect_ratio=(fraction)1/1 /pipeline0/queue0.sink: caps = video/x-raw-yuv, width=(int)404, height=(int)424, format=(fourcc)YUY2, framerate=(fraction)20/1, pixel_aspect_ratio=(fraction)1/1 /pipeline0/queue0.src: caps = video/x-raw-yuv, width=(int)404, height=(int)424, format=(fourcc)YUY2, framerate=(fraction)20/1, pixel_aspect_ratio=(fraction)1/1 /pipeline0/typefindelement0.src: caps = video/x-raw-yuv, width=(int)404, height=(int)424, format=(fourcc)YUY2, framerate=(fraction)20/1, pixel_aspect_ratio=(fraction)1/1 /pipeline0/typefindelement0.sink: caps = video/x-raw-yuv, width=(int)404, height=(int)424, format=(fourcc)YUY2, framerate=(fraction)20/1, pixel_aspect_ratio=(fraction)1/1 ERROR: from element /pipeline0/tcpserversrc0: Internal data flow error. Additional debug info: gstbasesrc.c(2193): gst_base_src_loop (): /pipeline0/tcpserversrc0: streaming task paused, reason not-negotiated (-4) ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... /pipeline0/typefindelement0.src: caps = NULL /pipeline0/typefindelement0.sink: caps = NULL /pipeline0/queue0.src: caps = NULL /pipeline0/queue0.sink: caps = NULL /pipeline0/splitter.src0: caps = NULL /pipeline0/splitter.sink: caps = NULL /pipeline0/videoparse0.src: caps = NULL FREEING pipeline ... Once this works, there will be another queue off of the 'splitter' tee that goes through some processing before heading out to a gsrtpbin/udpsink (hence the part about a gstpbin at the start of the pipeline. removing that part yeilds the same results). Thanks -Brian On Mon, Jan 12, 2009 at 10:33 AM, Arnout Vandecappelle wrote: > On Thursday 08 January 2009 22:48:04 OB Lutz wrote: > > I'm trying to write a pipeline that will take video in from a network > > source, do some transcoding, and send it out to a network sink. No > problem > > there. But I need to be able to get preview snapshots of the stream as > its > > running, either on demand, or just have it continuously dump out frames > to > > a known file periodically. I'm doing some experiments using a v4lsrc and > > xvimagesink as my fake network source and sink for now. I've found the > > 'tee' block that looks promising to split my stream into 2, so one can go > > on its way to the network sink, and the other could maybe write to a > file, > > but all my experiments have only shown me a single frame in my > xvimagesink. > > Am I going down the right path trying to use 'tee' or is there some other > > set of blocks I should use? > > You probably need to insert a queue behind the tee. Otherwise you can get > deadlocks because all sinks try to synchronize from the same thread. > > Failing that, can you post the gst-launch command you're using? > > Regards, > Arnout > > -- > Arnout Vandecappelle arnout at mind be > Senior Embedded Software Architect +32-16-286540 > Essensium/Mind http://www.mind.be > G.Geenslaan 9, 3001 Leuven, Belgium BE 872 984 063 RPR > Leuven > LinkedIn profile: http://www.linkedin.com/in/arnoutvandecappelle > GPG fingerprint: D206 D44B 5155 DF98 550D 3F2A 2213 88AA A1C7 C933 > > > ------------------------------------------------------------------------------ > Check out the new SourceForge.net Marketplace. > It is the best place to buy or sell services for > just about anything Open Source. > http://p.sf.net/sfu/Xq1LFB > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liangzhihong1984 at 126.com Tue Jan 13 04:19:59 2009 From: liangzhihong1984 at 126.com (liangzhihong1984) Date: Tue, 13 Jan 2009 11:19:59 +0800 (CST) Subject: [gst-devel] No "get-internal-session" in my gstrtpbin Message-ID: <29618967.97661231816799662.JavaMail.coremail@bj126app84.126.com> I got the information from GStreamer website about "get-internal-session" in gstrtpbin. However when I ran "gst-inspect gstrtpbin", there's no this signal at all. Was it replaced by something else? My gstreamer version is 0.10.8, can anybody give me some help. Thank you in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: From jpuydt at free.fr Tue Jan 13 09:03:28 2009 From: jpuydt at free.fr (Julien Puydt) Date: Tue, 13 Jan 2009 09:03:28 +0100 Subject: [gst-devel] I can't get an appsrc-using pipeline to PLAYING state Message-ID: <496C4AD0.10504@free.fr> Hi, now the app plugin has gone into base, I'm trying to get my defunct gstreamer code in ekiga to actually work. The first annoying thing I see is that my pipeline doesn't seem to go into the PLAYING state ; I was able to reproduce the problem pretty easily : LANG=C gst-launch appsrc caps=audio/x-raw-int,rate=44100,channels=2,width=16 name=ekiga_src ! volume name=ekiga_volume ! sdlaudiosink Setting pipeline to PAUSED ... Pipeline is PREROLLING ... and it never goes into PLAYING. Could someone hint what's going wrong there ? Snark on #gstreamer From paul at linuxaudiosystems.com Tue Jan 13 09:15:45 2009 From: paul at linuxaudiosystems.com (Paul Davis) Date: Tue, 13 Jan 2009 09:15:45 +0100 Subject: [gst-devel] VST Audio Plug in approach using Gstreamer and Linux OS with ARM Cortex A8 processors In-Reply-To: References: Message-ID: <1231834545.2685.65.camel@dhin> On Sat, 2009-01-10 at 10:10 -0800, thx1138 wrote: > > So, If I understand your response, VST plugins compiled for Windowx/MacOSX they are compiled for an OS *and* a CPU architecture: windows/x86 and OSX/ppc or OSX/x86. you cannot take these plugins an expect to run them on linux/arm. you could not even run them on windows/arm or windows/alpha (for example). > would need to be recompiled to run as plug-ins using Gstreamer. if you use an ARM processor, they would have to be recompiled one way or another even if you ran windows and use VST. > Wine just give a Windows emulation for x86 emulation I gather. not precisely, but close enough. the point is that the plugins will not run on a different CPU architecture without being recompiled. unless you have source code for the plugin, its a non-starter. > Our software manager has decided to build our Dolby, DTS, etc. plug-ins to > use Gstreamer as interface which I believe is fine, I just need other VST > applications to be ported to Gstreamer. from the sound of it you need VST *plugins* not VST-supporting host applications. > I wish I had some quick way to evaluate Gstreamer plug-ins versus VST > approach. Perhaps this will take some effort to gather benchmarks on > latency. the VST architecture has no impact on latency. the audio latency is determined by the host application, the plugins have no way to affect it. i do not know enough about gstreamer to understand its approach to latency, but i do know that at least as of a few months ago, gstreamer was not designed around the same principles as most low latency audio APIs (ASIO, CoreAudio, JACK). however, it may still be able to get down to the range you need - unless you are creating an application for pro-audio and/or musical performance, the acceptable range is pretty wide. From vaisakh.n at nestgroup.net Tue Jan 13 10:35:57 2009 From: vaisakh.n at nestgroup.net (vaisakh.n) Date: Tue, 13 Jan 2009 15:05:57 +0530 Subject: [gst-devel] Gst-editor-0.10 Message-ID: <496C607D.4030209@nestgroup.net> Hi I was trying to create a video pipeline in ./gst-editor (0.10) when played it its quits and got error message like this /*** gst-editor: cairo-surface.c:368: cairo_surface_reference: Assertion `surface->ref_count > 0' failed. **/ and the pipeline is /gst-launch-0.10 filesrc location=Test.mpg ! mpegdemux name=demux ! mpeg2dec ! queue ! ffmpegcsp ! xvimagesink demux. ! mad ! queue ! audioconvert ! osssink // help me.. thanks ***** Confidentiality Statement/Disclaimer ***** This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. From jyoti.d at allaboutif.com Tue Jan 13 10:40:46 2009 From: jyoti.d at allaboutif.com (Jyoti D) Date: Tue, 13 Jan 2009 15:10:46 +0530 Subject: [gst-devel] Regarding decodebin Message-ID: Hi All, My query is regarding usage of decodebin in the application program. I have been using decodebin and uridecodebin quite often while launching pipeline. But is it recommended to be used in application also? How much efficient would it be using in application? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From dstaples at friendmts.co.uk Tue Jan 13 12:15:04 2009 From: dstaples at friendmts.co.uk (Darren Staples) Date: Tue, 13 Jan 2009 11:15:04 +0000 Subject: [gst-devel] Message-ID: I?ve got my movie playing in a preview window, with an appsink element supplying buffers to my app. (thanks to your help !) :) It may be that I can process the frame buffers faster than normal playback rate, so putting the pipeline into PLAYING mode will actually slow down my application. What?s the best way to ensure I process each frame as fast as possible ? The two ways I can see are: 1. Use a g_main_loop and set up an Oidle? call-back function to process each frame, using g_idle_add(). Can I then seek to the next frame from within this idle call-back function ? 2. Scrap the g_main_loop and use a while( !gst_app_sink_is_eos( sink )) loop. To seek to the next frame presumably I need to do a gst_element_seek_simple() with GST_FORMAT_DEFAULT and the seek_pos set to the frame number ? Sorry to ask so many questions... Darren *** FRIEND MTS LIMITED EMAIL CONFIDENTIALITY *** This email message and any attachments may contain information which is confidential or privileged and is intended for the sole use of the person to whom it is addressed. If you are not the intended recipient, be aware that any disclosures, copying, distribution or use of the contents is prohibited. If you have received this email message in error, please notify our office by telephone (+44 (0)121 633 2000) or email (admin at friendmts.co.uk) immediately. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: From wim.taymans at gmail.com Tue Jan 13 13:37:17 2009 From: wim.taymans at gmail.com (Wim Taymans) Date: Tue, 13 Jan 2009 13:37:17 +0100 Subject: [gst-devel] In-Reply-To: References: Message-ID: <1231850237.29567.5.camel@metal> On Tue, 2009-01-13 at 11:15 +0000, Darren Staples wrote: > I?ve got my movie playing in a preview window, with an appsink element > supplying buffers to my app. (thanks to your help !) :) > > It may be that I can process the frame buffers faster than normal > playback rate, so putting the pipeline into PLAYING mode will actually > slow down my application. > What?s the best way to ensure I process each frame as fast as > possible ? > > The two ways I can see are: > 1. Use a g_main_loop and set up an ?idle? call-back function to > process each frame, using g_idle_add(). Can I then seek to the > next frame from within this idle call-back function ? > 2. Scrap the g_main_loop and use a while( ! > gst_app_sink_is_eos( sink )) loop. > > To seek to the next frame presumably I need to do a > gst_element_seek_simple() with GST_FORMAT_DEFAULT and the seek_pos set > to the frame number ? > Set the sync property on appsink to FALSE, this will disable synchronisation against the clock. Wim > Sorry to ask so many questions... > > Darren > > > > *** FRIEND MTS LIMITED EMAIL CONFIDENTIALITY *** > > This email message and any attachments may contain information which > is confidential or privileged and is intended for the sole use of the > person to whom it is addressed. If you are not the intended > recipient, be aware that any disclosures, copying, distribution or use > of the contents is prohibited. If you have received this email > message in error, please notify our office by telephone (+44 (0)121 > 633 2000) or email (admin at friendmts.co.uk) immediately. Thank you. > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From rahul.verma at sasken.com Tue Jan 13 12:43:35 2009 From: rahul.verma at sasken.com (Rahul Verma) Date: Tue, 13 Jan 2009 17:13:35 +0530 Subject: [gst-devel] 3gp demux plugin working on PC but not on target Message-ID: <89635EF2797FED4B9EF9769F80CEABC81AF3BD@EXGZ01.sasken.com> Hi, >Really, this doesn't help too much. You'll need to attach a debugger >and see where things are going wrong. It could be that one of the >libraries gstreamer uses was compiled wrongly, so things are going >wrong "randomly", or it could be a bug in your plugin - we can't >guess. You might also find running it on a PC using a tool like >valgrind helpful - it might show you something you're doing wrong that >just luckily doesn't cause a crash on the PC, but does on your target >system. Finally I was able to get Audio and Video working on target using my 3gp demux. As you pointed out correctly, bug was in my code. I was trying to access the file location from filesrc element using the following code, and it caused the crash. fileSrcPeerPad = gst_pad_get_peer(filter->sinkPad); if(NULL == fileSrcPeerPad) { GST_DEBUG("gone.. no src peer pad. "); } //extract the file src element here and get the filename, fd and uri fileSrcPeerElement = gst_pad_get_parent_element(fileSrcPeerPad); element_class = GST_ELEMENT_CLASS(fileSrcPeerElement); object_class = G_OBJECT_CLASS(element_class); //try to extract the location from this GObject. g_object_get(object_class, "location", &fileLocation, NULL); and after that do the file operations. Surprisingly, this piece of code worked on PC. Can you help me in finding the error in the code?? Also, now I have some AV Sync issues. Till now I have not implemented any AV Sync logic in my Demux or Sink. I am passing the video buffers decoded to the sink, but it seems that video frames are dropped by the framework and after that I see continuous breaks. Does the pipeline implement the frame drop logic? If yes what logic does it follow for frame drops? can you guide me to the piece of code in the framework for a better understanding. Regards, Rahul >> Hi, >> >> >>> That said, why aren't you using qtdemux? GStreamer includes a fairly >>> full-featured, high quality demuxer for >>> quicktime-and-qt-derived-formats like 3gp, and it works well already. >> >> We use the qtdemux extensively and find it very stable. We are developing >> out 3gp plugin as an experiment just to learn more about the GStreamer >> framework. >Fair enough. >> >>> Without the source to your plugin, all we can say about this >>> particular crash is "it's presumably a bug in your plugin, you'll need >>> to fix it". >Really, this doesn't help too much. You'll need to attach a debugger >and see where things are going wrong. It could be that one of the >libraries gstreamer uses was compiled wrongly, so things are going >wrong "randomly", or it could be a bug in your plugin - we can't >guess. You might also find running it on a PC using a tool like >valgrind helpful - it might show you something you're doing wrong that >just luckily doesn't cause a crash on the PC, but does on your target >system. SASKEN BUSINESS DISCLAIMER ------------------------- This message may contain confidential, proprietary or legally privileged information. 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URL: From dstaples at friendmts.co.uk Tue Jan 13 14:46:08 2009 From: dstaples at friendmts.co.uk (Darren Staples) Date: Tue, 13 Jan 2009 13:46:08 +0000 Subject: [gst-devel] Re appsink sync property Message-ID: I?m not sure what you are referring to Wim. Could you elaborate please ? Many thanks, Darren *** FRIEND MTS LIMITED EMAIL CONFIDENTIALITY *** This email message and any attachments may contain information which is confidential or privileged and is intended for the sole use of the person to whom it is addressed. If you are not the intended recipient, be aware that any disclosures, copying, distribution or use of the contents is prohibited. If you have received this email message in error, please notify our office by telephone (+44 (0)121 633 2000) or email (admin at friendmts.co.uk) immediately. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: From wim.taymans at gmail.com Tue Jan 13 16:06:52 2009 From: wim.taymans at gmail.com (Wim Taymans) Date: Tue, 13 Jan 2009 16:06:52 +0100 Subject: [gst-devel] Re appsink sync property In-Reply-To: References: Message-ID: Try: gst-inspect appsink It shows you that there is a property called 'sync' which you can set to FALSE like this: g_object_set (G_OBJECT (appsink), "sync", FALSE, NULL); Wim On Tue, Jan 13, 2009 at 2:46 PM, Darren Staples wrote: > I'm not sure what you are referring to Wim. > Could you elaborate please ? > > Many thanks, > > Darren > > > > *** FRIEND MTS LIMITED EMAIL CONFIDENTIALITY *** > > This email message and any attachments may contain information which is > confidential or privileged and is intended for the sole use of the person to > whom it is addressed. If you are not the intended recipient, be aware that > any disclosures, copying, distribution or use of the contents is > prohibited. If you have received this email message in error, please notify > our office by telephone (+44 (0)121 633 2000) or email > (admin at friendmts.co.uk) immediately. Thank you. > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From mapping at sheerman-chase.org.uk Tue Jan 13 18:21:38 2009 From: mapping at sheerman-chase.org.uk (mapping at sheerman-chase.org.uk) Date: Tue, 13 Jan 2009 17:21:38 +0000 Subject: [gst-devel] gdkpixbufsink factory not found Message-ID: <20090113172138.7oggdpv3k8sccw0s@maggie.enixns.com> I have a strange problem with a factory not being found for "gdkpixbufsink" on Fedora Core 6 - while the same code works on Ubuntu 8.10. When I do "locate libgstgdk" on Fedora Core 6, I get: /usr/lib/gstreamer-0.10/libgstgdkpixbuf.so /usr/lib/gstreamer-0.8/libgstgdkpixbuf.so which is fine. It was installed by the normal good plugins RPM package. When I run gst_element_factory_find ("gdkpixbufsink"), I get a null pointer. Is there any reason why the factory is not found despite the library is installed? Is there any way to list my plugins? Is it a possibility that 0.8 and 0.10 are interferring? Incidentially, filesrc, decodebin, queue and ffmpegcolorspace have factories available, as expected. Thanks, Tim PS My previous problem about getting accurate frames is still a problem but I worked around it by using the time in nanoseconds and the frame rate. From thx1138 at earthlink.net Tue Jan 13 18:54:12 2009 From: thx1138 at earthlink.net (thx1138) Date: Tue, 13 Jan 2009 09:54:12 -0800 Subject: [gst-devel] VST Audio Plug in approach using Gstreamer and Linux OS with ARM Cortex A8 processors In-Reply-To: <1231834545.2685.65.camel@dhin> Message-ID: On 1/13/09 12:15 AM, "Paul Davis" wrote: > On Sat, 2009-01-10 at 10:10 -0800, thx1138 wrote: >> >> So, If I understand your response, VST plugins compiled for Windowx/MacOSX > > they are compiled for an OS *and* a CPU architecture: windows/x86 and > OSX/ppc or OSX/x86. you cannot take these plugins an expect to run them > on linux/arm. you could not even run them on windows/arm or > windows/alpha (for example). > >> would need to be recompiled to run as plug-ins using Gstreamer. > > if you use an ARM processor, they would have to be recompiled one way or > another even if you ran windows and use VST. > >> Wine just give a Windows emulation for x86 emulation I gather. > > not precisely, but close enough. the point is that the plugins will not > run on a different CPU architecture without being recompiled. unless you > have source code for the plugin, its a non-starter. > >> Our software manager has decided to build our Dolby, DTS, etc. plug-ins to >> use Gstreamer as interface which I believe is fine, I just need other VST >> applications to be ported to Gstreamer. > > from the sound of it you need VST *plugins* not VST-supporting host > applications. > >> I wish I had some quick way to evaluate Gstreamer plug-ins versus VST >> approach. Perhaps this will take some effort to gather benchmarks on >> latency. > > the VST architecture has no impact on latency. the audio latency is > determined by the host application, the plugins have no way to affect > it. > > i do not know enough about gstreamer to understand its approach to > latency, but i do know that at least as of a few months ago, gstreamer > was not designed around the same principles as most low latency audio > APIs (ASIO, CoreAudio, JACK). however, it may still be able to get down > to the range you need - unless you are creating an application for > pro-audio and/or musical performance, the acceptable range is pretty > wide. > > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel Hello Paul, This Terry Shultz, I was wondering is we could have a chat by phone. I am based in Los Angeles and I was hoping we could talk a bit more on the path in my earlier emails. My Cell# is 949.254.1159 Best regards, Terry From msmith at xiph.org Tue Jan 13 18:57:39 2009 From: msmith at xiph.org (Michael Smith) Date: Tue, 13 Jan 2009 09:57:39 -0800 Subject: [gst-devel] Regarding decodebin In-Reply-To: References: Message-ID: <3c1737210901130957v5b027f5ai693cd7b65385d08a@mail.gmail.com> On Tue, Jan 13, 2009 at 1:40 AM, Jyoti D wrote: > Hi All, > > My query is regarding usage of decodebin in the application program. > I have been using decodebin and uridecodebin quite often while launching > pipeline. > > But is it recommended to be used in application also? > How much efficient would it be using in application? It is very highly recommended - every major gstreamer-based player I know of uses decodebin (or decodebin2, or uridecodebin). Doing it otherwise would be a bad idea, generally. There is no efficiency issue with doing this. Mike From msmith at xiph.org Tue Jan 13 19:06:56 2009 From: msmith at xiph.org (Michael Smith) Date: Tue, 13 Jan 2009 10:06:56 -0800 Subject: [gst-devel] 3gp demux plugin working on PC but not on target In-Reply-To: <89635EF2797FED4B9EF9769F80CEABC81AF3BD@EXGZ01.sasken.com> References: <89635EF2797FED4B9EF9769F80CEABC81AF3BD@EXGZ01.sasken.com> Message-ID: <3c1737210901131006v54dd17aek92454b4705af5bd0@mail.gmail.com> On Tue, Jan 13, 2009 at 3:43 AM, Rahul Verma wrote: > Hi, > >>Really, this doesn't help too much. You'll need to attach a debugger >>and see where things are going wrong. It could be that one of the >>libraries gstreamer uses was compiled wrongly, so things are going >>wrong "randomly", or it could be a bug in your plugin - we can't >>guess. You might also find running it on a PC using a tool like >>valgrind helpful - it might show you something you're doing wrong that >>just luckily doesn't cause a crash on the PC, but does on your target >>system. > > Finally I was able to get Audio and Video working on target using my 3gp > demux. As you pointed out correctly, bug was in my code. I was trying to > access the file location from filesrc element using the following code, and > it caused the crash. > > fileSrcPeerPad = gst_pad_get_peer(filter->sinkPad); > if(NULL == fileSrcPeerPad) > { > GST_DEBUG("gone.. no src peer pad. "); > } > //extract the file src element here and get the filename, fd and uri > fileSrcPeerElement = gst_pad_get_parent_element(fileSrcPeerPad); > > element_class = GST_ELEMENT_CLASS(fileSrcPeerElement); > > object_class = G_OBJECT_CLASS(element_class); > > //try to extract the location from this GObject. > g_object_get(object_class, "location", &fileLocation, NULL); > > and after that do the file operations. Surprisingly, this piece of code > worked > on PC. > > Can you help me in finding the error in the code?? The whole concept is badly broken. Your element should not be poking at the implementation details of its peer - there's no reason to expect that the peer even HAS a "location" property. So, you should stop doing that entirely. I didn't check the actual code to see if there was an obvious bug in the actual implementation, because it really doesn't matter. > > Also, now I have some AV Sync issues. Till now I have not implemented any AV > Sync > logic in my Demux or Sink. I am passing the video buffers decoded to the > sink, but it > seems that video frames are dropped by the framework and after that I see > continuous > breaks. Does the pipeline implement the frame drop logic? If yes what logic > does it follow > for frame drops? can you guide me to the piece of code in the framework for > a better > understanding. Your demuxer should not implement any sync logic - it should timestamp the buffers it outputs, but nothing more than that. The sink is responsible for all synchronisation - but in general, you shouldn't need to implement anything special. The base classes (GstBaseSink, or subclasses like GstBaseVideoSink) should implement everything for you - you just need to specify that it's a synchronised sink. Frame dropping is also generally done by the sink base classes, if the buffers arrive too late. Frame dropping SHOULD also be implemented in decoders (particularly video decoders) in response to QoS messages from the sinks. The documentation goes into more detail about how this works (can't remember precisely where, probably the design docs). Mike From lutz at topfrose.de Tue Jan 13 20:57:43 2009 From: lutz at topfrose.de (Lutz =?ISO-8859-1?Q?M=FCller?=) Date: Tue, 13 Jan 2009 20:57:43 +0100 Subject: [gst-devel] I can't get an appsrc-using pipeline to PLAYING state In-Reply-To: <496C4AD0.10504@free.fr> References: <496C4AD0.10504@free.fr> Message-ID: <1231876664.4079.2.camel@acer> On Tue, 2009-01-13 at 09:03 +0100, Julien Puydt wrote: > LANG=C gst-launch appsrc > caps=audio/x-raw-int,rate=44100,channels=2,width=16 name=ekiga_src ! > volume name=ekiga_volume ! sdlaudiosink > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > > and it never goes into PLAYING. > > Could someone hint what's going wrong there ? According to the documentation, "Sink elements only accept one single buffer of data and then block. At this point the pipeline is ?prerolled? and ready to render data immediately." I guess sdlaudiosink is waiting for this one single buffer in vain in your pipeline forever... Lutz From thaytan at noraisin.net Tue Jan 13 23:49:55 2009 From: thaytan at noraisin.net (Jan Schmidt) Date: Tue, 13 Jan 2009 22:49:55 +0000 Subject: [gst-devel] 3rd pre-releases of Core/Base available. Bad/Python unchanged. In-Reply-To: <1231544503.21410.4.camel@fancy-ubuntu> References: <1231279326.6539.5.camel@fancy-ubuntu> <1231286471.6539.17.camel@fancy-ubuntu> <1231544503.21410.4.camel@fancy-ubuntu> Message-ID: <1231886995.1445.6.camel@fancy-ubuntu> Hiyas, New pre-releeases of Core and Base are available at http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.21.4.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.21.4.tar.bz2 These fix a few cosmetic issues. gst-plugins-bad and gst-python are still at http://gstreamer.freedesktop.org/src/gst-plugins-bad/pre/gst-plugins-bad-0.10.9.3.tar.bz2 and http://gstreamer.freedesktop.org/src/gst-python/pre/gst-python-0.10.13.3.tar.bz2 The releases are due out Monday 19th January, unless new blocker bugs are found and filed. Cheers, Jan. On Fri, 2009-01-09 at 23:41 +0000, Jan Schmidt wrote: > Hi all, > > A new set of pre-releases of all 4 modules are now available at: > > http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.21.3.tar.bz2 > http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.21.3.tar.bz2 > http://gstreamer.freedesktop.org/src/gst-plugins-bad/pre/gst-plugins-bad-0.10.9.3.tar.bz2 > and > http://gstreamer.freedesktop.org/src/gst-python/pre/gst-python-0.10.13.3.tar.bz2 > > These fix a variety of small bugs and omissions. Please give them some > testing love. File bugs at http://bugzilla.gnome.org > > If necessary, there'll be new pre-releases on Tuesday (13 Jan). The > final release tarballs are expected on Monday week, the 19th January. > > After that, we'll be staying frozen for a bit while we switch all the > repositories over to git. > > Cheers, > Jan. > > On Wed, 2009-01-07 at 00:01 +0000, Jan Schmidt wrote: > > Hi all, > > > > The first pre-releases of the 4 modules are now available at: > > http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.21.2.tar.bz2 > > http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.21.2.tar.bz2 > > http://gstreamer.freedesktop.org/src/gst-plugins-bad/pre/gst-plugins-bad-0.10.9.2.tar.bz2 > > and > > http://gstreamer.freedesktop.org/src/gst-python/pre/gst-python-0.10.13.2.tar.bz2 > > > > Please test, and file any bugs at http://bugzilla.gnome.org/ ! > > > > Expect new pre-releases on Friday if needed. Check the release plan to > > find out about the release schedule: > > http://www.gstreamer.net/wiki/ReleasePlanning2009-1 > > > > Cheers, > > Jan. > > > > > > On Tue, 2009-01-06 at 22:02 +0000, Jan Schmidt wrote: > > > Hi all, > > > > > > All 4 of the Core, Base Plugins, Bad Plugins and Python bindings modules > > > are now frozen for releases. The Bad Plugins aren't normally in this > > > cycle. They're there this month to handle appsrc and appsink moving from > > > -bad to -base at last, and audioresample trading places with the much > > > better speexresample. Speexresample is now in -base, but registers with > > > the name 'audioresample'. The old audioresample is now in -bad as > > > 'legacyresample' > > > > > > See the release plan for the first few months of this year > > > (http://www.gstreamer.net/wiki/ReleasePlanning2009-1) for more detail, > > > and expect the pre-releases later tonight. > > > > > > Cheers, > > > Jan. -- Jan Schmidt From liangzhihong1984 at 126.com Wed Jan 14 06:29:51 2009 From: liangzhihong1984 at 126.com (liangzhihong1984) Date: Wed, 14 Jan 2009 13:29:51 +0800 (CST) Subject: [gst-devel] gstreamer-0.10.21 installation error Message-ID: <7927136.476191231910991237.JavaMail.coremail@bj126app87.126.com> When I tried to install gstreamer-0.10.21 on Ubuntu 7.10 Errors like this occured: -- Installing ./html/style.css -- Installing ./html/gstreamer.devhelp make[5]: *** [install-data-local] Error 1 make[5]: Leaving directory `/root/Desktop/GStreamer/gstreamer-0.10.21/docs/gst' make[4]: *** [install-am] Error 2 make[4]: Leaving directory `/root/Desktop/GStreamer/gstreamer-0.10.21/docs/gst' make[3]: *** [install] Error 2 make[3]: Leaving directory `/root/Desktop/GStreamer/gstreamer-0.10.21/docs/gst' make[2]: *** [install-recursive] Error 1 make[2]: Leaving directory `/root/Desktop/GStreamer/gstreamer-0.10.21/docs' make[1]: *** [install] Error 2 make[1]: Leaving directory `/root/Desktop/GStreamer/gstreamer-0.10.21/docs' make: *** [install-recursive] Error 1 Can anybody help me to resolve this problem? Thank you in advance. Best Regards, Leon Liang -------------- next part -------------- An HTML attachment was scrubbed... URL: From nicolas.m.zhang at gmail.com Wed Jan 14 08:36:13 2009 From: nicolas.m.zhang at gmail.com (Eric Zhang) Date: Wed, 14 Jan 2009 15:36:13 +0800 Subject: [gst-devel] Grabbing RTSP Stats? In-Reply-To: References: Message-ID: Hi, gstreamer: 1. RTCP does the communication works between server & clients. Also gstrtpbin handles the RTCP stuffs. 2. gstrtpjitterbuffer will send a customized event to downstream elements when it finds packet missing. You can make a packet receiving report according to this. Eric Zhang 2009/1/13 Jack Foster > Hiya Gang, > > I currently have an rtp/rtsp server -> client stream via gstrtpbin. > Is there a way I can get access to the statistics of the sender and > receiver reports? I'd like the server to know if the client missed > packets or if the communication path is fully broken. Would this be > the on-timeout signal? > > Jack > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From nicolas.m.zhang at gmail.com Wed Jan 14 08:41:26 2009 From: nicolas.m.zhang at gmail.com (Eric Zhang) Date: Wed, 14 Jan 2009 15:41:26 +0800 Subject: [gst-devel] No "get-internal-session" in my gstrtpbin In-Reply-To: <29618967.97661231816799662.JavaMail.coremail@bj126app84.126.com> References: <29618967.97661231816799662.JavaMail.coremail@bj126app84.126.com> Message-ID: Hi, gstreamer-devel: I think what you got from gstreamer website is obsolete. gstrtpbin has no signal "get-internal-session". Eric Zhang 2009/1/13 liangzhihong1984 > > I got the information from GStreamer website about "get-internal-session" > in gstrtpbin. > However when I ran "gst-inspect gstrtpbin", there's no this signal at > all. Was it replaced by something else? > My gstreamer version is 0.10.8, can anybody give me some help. > > Thank you in advance. > > > ------------------------------ > ???????????????????5000???? > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jpuydt at free.fr Wed Jan 14 09:15:12 2009 From: jpuydt at free.fr (Julien Puydt) Date: Wed, 14 Jan 2009 09:15:12 +0100 Subject: [gst-devel] I can't get an appsrc-using pipeline to PLAYING state In-Reply-To: <1231876664.4079.2.camel@acer> References: <496C4AD0.10504@free.fr> <1231876664.4079.2.camel@acer> Message-ID: <496D9F10.2020004@free.fr> Lutz M?ller a ?crit : > On Tue, 2009-01-13 at 09:03 +0100, Julien Puydt wrote: >> LANG=C gst-launch appsrc >> caps=audio/x-raw-int,rate=44100,channels=2,width=16 name=ekiga_src ! >> volume name=ekiga_volume ! sdlaudiosink >> Setting pipeline to PAUSED ... >> Pipeline is PREROLLING ... >> >> and it never goes into PLAYING. >> >> Could someone hint what's going wrong there ? > > According to the documentation, "Sink elements only accept one single > buffer of data and then block. At this point the pipeline is ?prerolled? > and ready to render data immediately." I guess sdlaudiosink is waiting > for this one single buffer in vain in your pipeline forever... Hmmmm... when I modified my code to go on even with a non-PLAYING pipeline, I see the buffers getting pushed (printf in the pushing function), but I don't hear anything :-/ How can I discover what happens within the pipeline? Thanks, Snark From arnout at mind.be Wed Jan 14 09:15:17 2009 From: arnout at mind.be (Arnout Vandecappelle) Date: Wed, 14 Jan 2009 09:15:17 +0100 Subject: [gst-devel] Grabbing snapshots out of live stream In-Reply-To: References: <200901121633.15781.arnout@mind.be> Message-ID: <200901140915.18851.arnout@mind.be> On Monday 12 January 2009 21:24:20 OB Lutz wrote: > gst-launch -vvv gstrtpbin name=rtpbin \ > ????????????? tcpserversrc host=127.0.0.1 port=1235 ! videoparse > format=YUY2 width=404 height=424 framerate=20/1 bpp=16 ! \ tee > name=splitter splitter.? ! queue ! typefind ! pngenc snapshot=false ! > multifilesink sync=false preroll-queue-len=20 location=/tmp/output.png -vvv > splitter. You don't need a typefind in your pipeline. It fails to find the type of the raw video (there's nothing in a raw video stream to indicate what type it is) and therefore it doesn't negotiate its caps with the pngenc. To debug a pipeline, it's convenient to strip it down until something works, that way you can find out which element causes the problem. You can replace a video source with videotestsrc and the sink with fakesink. That way, you'll find out that videotestsrc ! typefind ! pngenc snapshot=false ! fakesink doesn't work, while videotestsrc ! pngenc snapshot=false ! fakesink does work. Regards, Arnout -- Arnout Vandecappelle arnout at mind be Senior Embedded Software Architect +32-16-286540 Essensium/Mind http://www.mind.be G.Geenslaan 9, 3001 Leuven, Belgium BE 872 984 063 RPR Leuven LinkedIn profile: http://www.linkedin.com/in/arnoutvandecappelle GPG fingerprint: D206 D44B 5155 DF98 550D 3F2A 2213 88AA A1C7 C933 From mapping at sheerman-chase.org.uk Wed Jan 14 10:49:39 2009 From: mapping at sheerman-chase.org.uk (mapping at sheerman-chase.org.uk) Date: Wed, 14 Jan 2009 09:49:39 +0000 Subject: [gst-devel] gdkpixbufsink factory not found Message-ID: <20090114094939.u964aro9w0ckww4s@maggie.enixns.com> Hey, A quick follow up, in which I have found my problem but not a solution. Going gst-inspect gdkpixbuf shows I don't have the gdkpixbufsink factory plugin in my version of gdkpixbuf (0.10.4). My next step is to investigate if using gdkpixbufdec is a possibility instead of gdkpixbufsink. Or possibly use a different pixel buffer (if any exist). Do I need to have a sink element after gdkpixbufdec? If so, which one? Is there any logical way to indicate when the buffer is updated? (basically to reproduce gdkpixbufsink's behaviour) Thanks, Tim From wim.taymans at gmail.com Wed Jan 14 11:53:09 2009 From: wim.taymans at gmail.com (Wim Taymans) Date: Wed, 14 Jan 2009 11:53:09 +0100 Subject: [gst-devel] No "get-internal-session" in my gstrtpbin In-Reply-To: <29618967.97661231816799662.JavaMail.coremail@bj126app84.126.com> References: <29618967.97661231816799662.JavaMail.coremail@bj126app84.126.com> Message-ID: <1231930389.19973.29.camel@metal> On Tue, 2009-01-13 at 11:19 +0800, liangzhihong1984 wrote: > > I got the information from GStreamer website about > "get-internal-session" in gstrtpbin. > However when I ran "gst-inspect gstrtpbin", there's no this signal > at all. Was it replaced by something else? > My gstreamer version is 0.10.8, can anybody give me some help. > Your version of gst-plugins-bad is way too old, you need CVS or wait for version 0.10.10 that will be released in a couple of days. Wim > Thank you in advance. > > > > > ______________________________________________________________________ > ???????????????????5000???? > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From amm at tid.es Wed Jan 14 12:08:38 2009 From: amm at tid.es (Antonio =?ISO-8859-1?Q?Marqu=E9s?=) Date: Wed, 14 Jan 2009 12:08:38 +0100 Subject: [gst-devel] plugin working with realimentation Message-ID: <1231931318.5877.742.camel@viladecans> Hello developers, I am interesting to design a plugin which has as inputs N-1 number of pads (some images from a filesrc for example) and another input which would be the result of the plugin's output, so I would like to implement a kind of realimentation in that plugin. In that plugin I would like to have all the N input buffers in the process function. So I have thought in using the collect pad method for all the pads. The problem is that, for the first buffer I would have not the buffer of the pad N (because it would be the output of the buffer), so the pipeline would not be set on PLAYING state never. Does anyone know how to solve this problem? The main purpose of the plugin is, to make an initial computation for the first frame taking into account the N-1 images and for the other frames taking also into account the previous output, which would be the N pad. If anyone knows a plugin with similar function it would be great for me to look at its source code. Thanks in advance. -- Toni Marqu?s Marqu?s Telef?nica I+D Divisi?n de Tecnolog?as de Video amm at tid.es 933653188 From vaisakh.n at nestgroup.net Wed Jan 14 14:18:01 2009 From: vaisakh.n at nestgroup.net (Vaisakh N.) Date: Wed, 14 Jan 2009 18:48:01 +0530 Subject: [gst-devel] mad plugin Message-ID: <245245F712CBF74D865D90E99497849F780B44@mail-3.tvm.nestgroup.net> hi all, can any body tell through which API gstreamer aceess the libgstmad.so.... help me thanks ***** Confidentiality Statement/Disclaimer ***** This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ob.lutz at gmail.com Wed Jan 14 15:47:51 2009 From: ob.lutz at gmail.com (OB Lutz) Date: Wed, 14 Jan 2009 09:47:51 -0500 Subject: [gst-devel] Grabbing snapshots out of live stream In-Reply-To: <200901140915.18851.arnout@mind.be> References: <200901121633.15781.arnout@mind.be> <200901140915.18851.arnout@mind.be> Message-ID: Thanks! On Wed, Jan 14, 2009 at 3:15 AM, Arnout Vandecappelle wrote: > On Monday 12 January 2009 21:24:20 OB Lutz wrote: > > gst-launch -vvv gstrtpbin name=rtpbin \ > > tcpserversrc host=127.0.0.1 port=1235 ! videoparse > > format=YUY2 width=404 height=424 framerate=20/1 bpp=16 ! \ tee > > name=splitter splitter. ! queue ! typefind ! pngenc snapshot=false ! > > multifilesink sync=false preroll-queue-len=20 location=/tmp/output.png > -vvv > > splitter. > > You don't need a typefind in your pipeline. It fails to find the type of > the > raw video (there's nothing in a raw video stream to indicate what type it > is) > and therefore it doesn't negotiate its caps with the pngenc. > > To debug a pipeline, it's convenient to strip it down until something > works, > that way you can find out which element causes the problem. You can > replace > a video source with videotestsrc and the sink with fakesink. That way, > you'll find out that > > videotestsrc ! typefind ! pngenc snapshot=false ! fakesink > > doesn't work, while > > videotestsrc ! pngenc snapshot=false ! fakesink > > does work. > > Regards, > Arnout > -- > Arnout Vandecappelle arnout at mind be > Senior Embedded Software Architect +32-16-286540 > Essensium/Mind http://www.mind.be > G.Geenslaan 9, 3001 Leuven, Belgium BE 872 984 063 RPR > Leuven > LinkedIn profile: http://www.linkedin.com/in/arnoutvandecappelle > GPG fingerprint: D206 D44B 5155 DF98 550D 3F2A 2213 88AA A1C7 C933 > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From r_poppi at gmx.at Wed Jan 14 16:41:16 2009 From: r_poppi at gmx.at (Roland Poppenreiter) Date: Wed, 14 Jan 2009 16:41:16 +0100 Subject: [gst-devel] libmpg123 decoder plugin Message-ID: <20090114154116.GZ14461@burning-bastards.com> Hello everybody, I hope this is the right mailinglist for this anouncement. I wrote a mp3 decoder plugin with the mpg123 library. One reason is that mpg123 has some processor optimization (SSE, MMX, 3DNow, ...) in it and should be faster than the mad decoder. I used the mad decoder plugin (gstmad.c) as template for my plugin - so they are quite similar. I wanted to benchmark both decoders and compare the results but I ran into several problems and I got a few questions: - sometimes i get a short white noise sound when i start playing an mp3 file with my mpg123 decoder plugin. - short white noise sound appears after seeking as well. - is it possible that data comming from the filesrc is sometimes overlapping? - seeking is not always working, also not with the mad decoder. - the mad decoder decodes mpg samples to audio/x-raw-int 32 Bit, but most extensions (SSE, MMX, ...) can just handle 16 Bit samples, therefore I would preferre 16 Bit audio/x-raw-int as output for my decoder. I wrote a small mediaplayer (described in the GStreamer Application Development Manual) to test my plugin. The problems occur with my mediaplayer and also with the gst-launch program. Sourcecode can be found here: https://semsemilla.kastel.net/gst/ I would appreciate little help to solve my problems or any opinion about this plugin. -- Roland Poppenreiter From wim.taymans at gmail.com Wed Jan 14 18:33:40 2009 From: wim.taymans at gmail.com (Wim Taymans) Date: Wed, 14 Jan 2009 18:33:40 +0100 Subject: [gst-devel] mad plugin In-Reply-To: <245245F712CBF74D865D90E99497849F780B44@mail-3.tvm.nestgroup.net> References: <245245F712CBF74D865D90E99497849F780B44@mail-3.tvm.nestgroup.net> Message-ID: I don't understand your question, what exactly do you want to know? Wim On Wed, Jan 14, 2009 at 2:18 PM, Vaisakh N. wrote: > > hi all, > can any body tell through which API gstreamer aceess the libgstmad.so.... > help me > thanks > > ***** Confidentiality Statement/Disclaimer ***** > > This message and any attachments is intended for the sole use of the > intended recipient. It may contain confidential information. Any > unauthorized use, dissemination or modification is strictly prohibited. If > you are not the intended recipient, please notify the sender immediately > then delete it from all your systems, and do not copy, use or print. > Internet communications are not secure and it is the responsibility of the > recipient to make sure that it is virus/malicious code exempt. > The company/sender cannot be responsible for any unauthorized alterations or > modifications made to the contents. If you require any form of confirmation > of the contents, please contact the company/sender. The company/sender is > not liable for any errors or omissions in the content of this message. > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From jpuydt at free.fr Wed Jan 14 21:12:15 2009 From: jpuydt at free.fr (Julien Puydt) Date: Wed, 14 Jan 2009 21:12:15 +0100 Subject: [gst-devel] I can't get an appsrc-using pipeline to PLAYING state In-Reply-To: <496D9F10.2020004@free.fr> References: <496C4AD0.10504@free.fr> <1231876664.4079.2.camel@acer> <496D9F10.2020004@free.fr> Message-ID: <496E471F.6040200@free.fr> Julien Puydt a ?crit : > How can I discover what happens within the pipeline? I added some more debug printing : that pipeline is in the PAUSED state, not the PLAYING state... that is why I don't hear anything :-( Any hint why a pipeline which was ordered : gst_element_set_state (pipeline[ii], GST_STATE_PLAYING); doesn't go to PLAYING but gets stuck in PAUSED would help :-( Thanks, Snark on #gstreamer From jpuydt at free.fr Wed Jan 14 21:19:21 2009 From: jpuydt at free.fr (Julien Puydt) Date: Wed, 14 Jan 2009 21:19:21 +0100 Subject: [gst-devel] I can't get an appsrc-using pipeline to PLAYING state In-Reply-To: <496C4AD0.10504@free.fr> References: <496C4AD0.10504@free.fr> Message-ID: <496E48C9.7070106@free.fr> Julien Puydt a ?crit : > The first annoying thing I see is that my pipeline doesn't seem to go > into the PLAYING state ; I was able to reproduce the problem pretty easily : > LANG=C gst-launch appsrc > caps=audio/x-raw-int,rate=44100,channels=2,width=16 name=ekiga_src ! > volume name=ekiga_volume ! sdlaudiosink I notice that if I had "is-live=true" for appsrc, then that pipeline goes into PLAYING more (not stuck on PREROLL)... But it doesn't solve my issue in ekiga, where the pipeline stays as PAUSED... Snark on #gstreamer From jpuydt at free.fr Wed Jan 14 21:57:58 2009 From: jpuydt at free.fr (Julien Puydt) Date: Wed, 14 Jan 2009 21:57:58 +0100 Subject: [gst-devel] I can't get an appsrc-using pipeline to PLAYING state In-Reply-To: <496E48C9.7070106@free.fr> References: <496C4AD0.10504@free.fr> <496E48C9.7070106@free.fr> Message-ID: <496E51D6.6020109@free.fr> Julien Puydt a ?crit : > Julien Puydt a ?crit : >> The first annoying thing I see is that my pipeline doesn't seem to go >> into the PLAYING state ; I was able to reproduce the problem pretty easily : >> LANG=C gst-launch appsrc >> caps=audio/x-raw-int,rate=44100,channels=2,width=16 name=ekiga_src ! >> volume name=ekiga_volume ! sdlaudiosink > I notice that if I had "is-live=true" for appsrc, then that pipeline > goes into PLAYING more (not stuck on PREROLL)... > > But it doesn't solve my issue in ekiga, where the pipeline stays as > PAUSED... Sigh... the pipeline missed a " ! " -- now it goes into PLAYING state, I see buffers getting pushed to it... but no sound. How can I debug the situation? Snark on #gstreamer From vaisakh.n at nestgroup.net Thu Jan 15 04:43:07 2009 From: vaisakh.n at nestgroup.net (Vaisakh N.) Date: Thu, 15 Jan 2009 09:13:07 +0530 Subject: [gst-devel] Gstreamer - plugin.. Message-ID: <245245F712CBF74D865D90E99497849F8DEC0F@mail-3.tvm.nestgroup.net> Can any one tell me what are API's available in GStreamer for writing a fresh plug-in (example: an mp3 codec ) Help me.. Thanks in advance... ***** Confidentiality Statement/Disclaimer ***** This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. -------------- next part -------------- An HTML attachment was scrubbed... URL: From jyoti.d at allaboutif.com Thu Jan 15 05:23:37 2009 From: jyoti.d at allaboutif.com (Jyoti D) Date: Thu, 15 Jan 2009 09:53:37 +0530 Subject: [gst-devel] Regarding decodebin In-Reply-To: <3c1737210901130957v5b027f5ai693cd7b65385d08a@mail.gmail.com> References: <3c1737210901130957v5b027f5ai693cd7b65385d08a@mail.gmail.com> Message-ID: Thanks.. On Tue, Jan 13, 2009 at 11:27 PM, Michael Smith wrote: > On Tue, Jan 13, 2009 at 1:40 AM, Jyoti D wrote: > > Hi All, > > > > My query is regarding usage of decodebin in the application program. > > I have been using decodebin and uridecodebin quite often while launching > > pipeline. > > > > But is it recommended to be used in application also? > > How much efficient would it be using in application? > > It is very highly recommended - every major gstreamer-based player I > know of uses decodebin (or decodebin2, or uridecodebin). Doing it > otherwise would be a bad idea, generally. > > There is no efficiency issue with doing this. > > Mike > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From felipe.contreras at gmail.com Thu Jan 15 06:47:20 2009 From: felipe.contreras at gmail.com (Felipe Contreras) Date: Thu, 15 Jan 2009 07:47:20 +0200 Subject: [gst-devel] libmpg123 decoder plugin In-Reply-To: <20090114154116.GZ14461@burning-bastards.com> References: <20090114154116.GZ14461@burning-bastards.com> Message-ID: <94a0d4530901142147w600c113fu1b5d2cc5844b4929@mail.gmail.com> On Wed, Jan 14, 2009 at 5:41 PM, Roland Poppenreiter wrote: > Hello everybody, > > I hope this is the right mailinglist for this anouncement. I wrote > a mp3 decoder plugin with the mpg123 library. One reason is that > mpg123 has some processor optimization (SSE, MMX, 3DNow, ...) in it > and should be faster than the mad decoder. I used the mad decoder > plugin (gstmad.c) as template for my plugin - so they are quite > similar. I wanted to benchmark both decoders and compare the results > but I ran into several problems and I got a few questions: > > - sometimes i get a short white noise sound when i start playing an > mp3 file with my mpg123 decoder plugin. > - short white noise sound appears after seeking as well. > - is it possible that data comming from the filesrc is sometimes > overlapping? > - seeking is not always working, also not with the mad decoder. Are you using the mp3parse element? > - the mad decoder decodes mpg samples to audio/x-raw-int 32 Bit, > but most extensions (SSE, MMX, ...) can just handle 16 Bit > samples, therefore I would preferre 16 Bit audio/x-raw-int as > output for my decoder. Then output that. Is that mpg123 library new? If I recall correctly the last time I tried it wasn't there, I even suggested them to create a library, but they didn't look too interested. Another option is to provide an OpenMAX IL API for the library. In that case the gst-openmax wrapper would be able to use the decoder transparently. -- Felipe Contreras From jpuydt at free.fr Thu Jan 15 07:00:30 2009 From: jpuydt at free.fr (Julien Puydt) Date: Thu, 15 Jan 2009 07:00:30 +0100 Subject: [gst-devel] I can't get an appsrc-using pipeline to PLAYING state In-Reply-To: <496E51D6.6020109@free.fr> References: <496C4AD0.10504@free.fr> <496E48C9.7070106@free.fr> <496E51D6.6020109@free.fr> Message-ID: <496ED0FE.4070003@free.fr> Julien Puydt a ?crit : > Sigh... the pipeline missed a " ! " -- now it goes into PLAYING state, I > see buffers getting pushed to it... but no sound. I replaced the sdlaudiosink with a filesink : empty file -- the buffers don't get through! Snark on #gstreamer From vaisakh.n at nestgroup.net Thu Jan 15 08:10:47 2009 From: vaisakh.n at nestgroup.net (vaisakh.n) Date: Thu, 15 Jan 2009 12:40:47 +0530 Subject: [gst-devel] gstreamer- video-pgm Message-ID: <496EE177.5000009@nestgroup.net> I was trying buil an application to play " mpg " file so that in program i created elements are ... filesrc location=f1.mpg , mpegdemux, mpeg2dec , xvimagesink are the elements i created .. by using gst_element_factory_make ("", ""); and linked the elements like this ..... if(! gst_element_link_many(Source,MPG_Demux, MPG_Decod,Video_Sink ,NULL) ) { /* but its goes here */ /* error in linking * / } i found that error is in linking in between MPG_Demux and MPG_Decod .. why its so??? help me thanks in advance ***** Confidentiality Statement/Disclaimer ***** This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. From dstaples at friendmts.co.uk Thu Jan 15 10:15:31 2009 From: dstaples at friendmts.co.uk (Darren Staples) Date: Thu, 15 Jan 2009 09:15:31 +0000 Subject: [gst-devel] Missing plugin for HTTP Message-ID: Hi, I?ve got the latest versions of the base, good, bad and ugly plugins installed, and am trying to get this very simple pipeline to work: gst-launch playbin uri=http://........... But I get the error OA HTTP protocol plugin is required to play this stream, but not installed?. Is there some other set of plugins I have missed ? I have a similar error for RTMP links also... Thanks in advance ! Darren *** FRIEND MTS LIMITED EMAIL CONFIDENTIALITY *** This email message and any attachments may contain information which is confidential or privileged and is intended for the sole use of the person to whom it is addressed. If you are not the intended recipient, be aware that any disclosures, copying, distribution or use of the contents is prohibited. If you have received this email message in error, please notify our office by telephone (+44 (0)121 633 2000) or email (admin at friendmts.co.uk) immediately. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: From nicolas.m.zhang at gmail.com Thu Jan 15 10:44:52 2009 From: nicolas.m.zhang at gmail.com (Eric Zhang) Date: Thu, 15 Jan 2009 17:44:52 +0800 Subject: [gst-devel] Gstreamer - plugin.. In-Reply-To: <245245F712CBF74D865D90E99497849F8DEC0F@mail-3.tvm.nestgroup.net> References: <245245F712CBF74D865D90E99497849F8DEC0F@mail-3.tvm.nestgroup.net> Message-ID: Hi, gstreamer-devel: If you mean how to write a gstreamer plugin, the Plugin writer's Guide is what you need. Eric Zhang 2009/1/15 Vaisakh N. > Can any one tell me what are API's available in GStreamer for writing a > fresh plug-in (example: an mp3 codec ) > > > > > > Help me.. > > > > Thanks in advance? > > > ***** Confidentiality Statement/Disclaimer ***** > > This message and any attachments is intended for the sole use of the > intended recipient. It may contain confidential information. Any > unauthorized use, dissemination or modification is strictly prohibited. If > you are not the intended recipient, please notify the sender immediately > then delete it from all your systems, and do not copy, use or print. > Internet communications are not secure and it is the responsibility of the > recipient to make sure that it is virus/malicious code exempt. > The company/sender cannot be responsible for any unauthorized alterations > or modifications made to the contents. If you require any form of > confirmation of the contents, please contact the company/sender. The > company/sender is not liable for any errors or omissions in the content of > this message. > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From nicolas.m.zhang at gmail.com Thu Jan 15 10:47:50 2009 From: nicolas.m.zhang at gmail.com (Eric Zhang) Date: Thu, 15 Jan 2009 17:47:50 +0800 Subject: [gst-devel] gstreamer- video-pgm In-Reply-To: <496EE177.5000009@nestgroup.net> References: <496EE177.5000009@nestgroup.net> Message-ID: Hi, gstreamer-devel: The mpegdemux has dynamic pads which you can't link to decoder directly. You need to read the gstreamer application develop manual for details. Eric Zhang 2009/1/15 vaisakh.n > I was trying buil an application to play " mpg " file > > so that in program i created elements > are ... filesrc location=f1.mpg , mpegdemux, mpeg2dec , xvimagesink > > are the elements i created .. by using gst_element_factory_make ("", ""); > and linked the elements like this ..... > if(! gst_element_link_many(Source,MPG_Demux, MPG_Decod,Video_Sink ,NULL) ) > { > /* but its goes here */ > /* error in linking * / > } > > i found that error is in linking in between MPG_Demux and MPG_Decod .. > why its so??? > > help me > > thanks in advance > > > > ***** Confidentiality Statement/Disclaimer ***** > > This message and any attachments is intended for the sole use of the > intended recipient. It may contain confidential information. Any > unauthorized use, dissemination or modification is strictly prohibited. If > you are not the intended recipient, please notify the sender immediately > then delete it from all your systems, and do not copy, use or print. > Internet communications are not secure and it is the responsibility of the > recipient to make sure that it is virus/malicious code exempt. > The company/sender cannot be responsible for any unauthorized alterations > or modifications made to the contents. If you require any form of > confirmation of the contents, please contact the company/sender. The > company/sender is not liable for any errors or omissions in the content of > this message. > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From nicolas.m.zhang at gmail.com Thu Jan 15 10:51:11 2009 From: nicolas.m.zhang at gmail.com (Eric Zhang) Date: Thu, 15 Jan 2009 17:51:11 +0800 Subject: [gst-devel] Missing plugin for HTTP In-Reply-To: References: Message-ID: Hi, gstreamer-devel: According to the gstreamer plugin list, seems the http support in gstreamer is based on neon. So install neon first then reinstall your gst-plugins-bad package. Eric Zhang 2009/1/15 Darren Staples > Hi, > > I've got the latest versions of the base, good, bad and ugly plugins > installed, and am trying to get this very simple pipeline to work: > gst-launch playbin uri=http://........... > > But I get the error 'A HTTP protocol plugin is required to play this > stream, but not installed'. > > Is there some other set of plugins I have missed ? > > I have a similar error for RTMP links also... > > Thanks in advance ! > > Darren > > > > *** FRIEND MTS LIMITED EMAIL CONFIDENTIALITY *** > > This email message and any attachments may contain information which is > confidential or privileged and is intended for the sole use of the person to > whom it is addressed. If you are not the intended recipient, be aware that > any disclosures, copying, distribution or use of the contents is > prohibited. If you have received this email message in error, please notify > our office by telephone (+44 (0)121 633 2000) or email ( > admin at friendmts.co.uk) immediately. Thank you. > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From thijsvermeir at gmail.com Thu Jan 15 10:58:06 2009 From: thijsvermeir at gmail.com (Thijs Vermeir) Date: Thu, 15 Jan 2009 10:58:06 +0100 Subject: [gst-devel] gstreamer- video-pgm In-Reply-To: <496EE177.5000009@nestgroup.net> References: <496EE177.5000009@nestgroup.net> Message-ID: <20090115095806.GA11927@TVFluendoHome> Hi, On Thu, Jan 15, 2009 at 12:40:47PM +0530, vaisakh.n wrote: > I was trying buil an application to play " mpg " file > > so that in program i created elements > are ... filesrc location=f1.mpg , mpegdemux, mpeg2dec , xvimagesink > > are the elements i created .. by using gst_element_factory_make ("", ""); > and linked the elements like this ..... > if(! gst_element_link_many(Source,MPG_Demux, MPG_Decod,Video_Sink ,NULL) ) > { > /* but its goes here */ > /* error in linking * / > } > > i found that error is in linking in between MPG_Demux and MPG_Decod .. > why its so??? Demuxers have most of the time dynamic pads. You need to link them in a callback function and not with gst_element_link(_many), like in the hello world example [1]. Gr, Thijs [1] http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/chapter-helloworld.html > > help me > > thanks in advance > > > > ***** Confidentiality Statement/Disclaimer ***** > > This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. > The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From ensonic at hora-obscura.de Thu Jan 15 13:40:46 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Thu, 15 Jan 2009 14:40:46 +0200 Subject: [gst-devel] Missing plugin for HTTP In-Reply-To: References: Message-ID: <496F2ECE.4010700@hora-obscura.de> Darren Staples schrieb: > Hi, > > I?ve got the latest versions of the base, good, bad and ugly plugins > installed, and am trying to get this very simple pipeline to work: > gst-launch playbin uri=http://........... > > But I get the error ?A HTTP protocol plugin is required to play this > stream, but not installed?. gnomevfs, gio or neon can do it. Stefan > > Is there some other set of plugins I have missed ? > > I have a similar error for RTMP links also... > > Thanks in advance ! > > Darren > > > > *** FRIEND MTS LIMITED EMAIL CONFIDENTIALITY *** > > This email message and any attachments may contain information which > is confidential or privileged and is intended for the sole use of the > person to whom it is addressed. If you are not the intended > recipient, be aware that any disclosures, copying, distribution or use > of the contents is prohibited. If you have received this email > message in error, please notify our office by telephone (+44 (0)121 > 633 2000) or email (admin at friendmts.co.uk > ) immediately. Thank you. > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From ensonic at hora-obscura.de Thu Jan 15 13:49:54 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Thu, 15 Jan 2009 14:49:54 +0200 Subject: [gst-devel] gdkpixbufsink factory not found In-Reply-To: <20090114094939.u964aro9w0ckww4s@maggie.enixns.com> References: <20090114094939.u964aro9w0ckww4s@maggie.enixns.com> Message-ID: <496F30F2.1090501@hora-obscura.de> mapping at sheerman-chase.org.uk schrieb: > Hey, > > A quick follow up, in which I have found my problem but not a > solution. Going gst-inspect gdkpixbuf shows I don't have the > gdkpixbufsink factory plugin in my version of gdkpixbuf (0.10.4). > > My next step is to investigate if using gdkpixbufdec is a possibility > instead of gdkpixbufsink. Or possibly use a different pixel buffer (if > any exist). Do I need to have a sink element after gdkpixbufdec? If > so, which one? > gdkpixbufdec is just a imagedecoder using gdkpixbuf library. You need a sink afterwards to render the image (e.g. ximagesink) Stefan > Is there any logical way to indicate when the buffer is updated? > (basically to reproduce gdkpixbufsink's behaviour) > > Thanks, > > Tim > > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From r_poppi at gmx.at Thu Jan 15 13:35:21 2009 From: r_poppi at gmx.at (Roland Poppenreiter) Date: Thu, 15 Jan 2009 13:35:21 +0100 Subject: [gst-devel] libmpg123 decoder plugin In-Reply-To: <94a0d4530901142147w600c113fu1b5d2cc5844b4929@mail.gmail.com> References: <20090114154116.GZ14461@burning-bastards.com> <94a0d4530901142147w600c113fu1b5d2cc5844b4929@mail.gmail.com> Message-ID: <20090115123521.GD14461@burning-bastards.com> * On Thu, 15 Jan 2009 at 7:47 +0200, Felipe Contreras wrote: > On Wed, Jan 14, 2009 at 5:41 PM, Roland Poppenreiter wrote: > > Hello everybody, > > > > I hope this is the right mailinglist for this anouncement. I wrote > > a mp3 decoder plugin with the mpg123 library. One reason is that > > mpg123 has some processor optimization (SSE, MMX, 3DNow, ...) in it > > and should be faster than the mad decoder. I used the mad decoder > > plugin (gstmad.c) as template for my plugin - so they are quite > > similar. I wanted to benchmark both decoders and compare the results > > but I ran into several problems and I got a few questions: > > > > - sometimes i get a short white noise sound when i start playing an > > mp3 file with my mpg123 decoder plugin. > > - short white noise sound appears after seeking as well. > > - is it possible that data comming from the filesrc is sometimes > > overlapping? > > - seeking is not always working, also not with the mad decoder. > > Are you using the mp3parse element? no, I dont even know this plugin .... the pipeline I'm using is this: gst-launch filesrc location="foo.mp3" ! mpg123 ! audioconvert ! osssink > > - the mad decoder decodes mpg samples to audio/x-raw-int 32 Bit, > > but most extensions (SSE, MMX, ...) can just handle 16 Bit > > samples, therefore I would preferre 16 Bit audio/x-raw-int as > > output for my decoder. > > Then output that. okay, fine! > Is that mpg123 library new? If I recall correctly the last time I > tried it wasn't there, I even suggested them to create a library, but > they didn't look too interested. the libmpg123 is not new, its the api from the mpg123 mp3-player (http://www.mpg123.de/api/) and I already used it for my mpg123 gstreamer plugin. > Another option is to provide an OpenMAX IL API for the library. In > that case the gst-openmax wrapper would be able to use the decoder > transparently. I dont know this library, I will take a look at it, but for the moment I have to solve my problems with the mpg123 plugin. > -- > Felipe Contreras > -- Roland Poppenreiter From ensonic at hora-obscura.de Thu Jan 15 13:47:30 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Thu, 15 Jan 2009 14:47:30 +0200 Subject: [gst-devel] gstreamer-0.10.21 installation error In-Reply-To: <7927136.476191231910991237.JavaMail.coremail@bj126app87.126.com> References: <7927136.476191231910991237.JavaMail.coremail@bj126app87.126.com> Message-ID: <496F3062.8090308@hora-obscura.de> hi, this is covered in the archive. Short answare - install gtk-doc as a workaround. Stefan liangzhihong1984 schrieb: > When I tried to install gstreamer-0.10.21 on Ubuntu 7.10 > Errors like this occured: > -- Installing ./html/style.css > -- Installing ./html/gstreamer.devhelp > make[5]: *** [install-data-local] Error 1 > make[5]: Leaving directory > `/root/Desktop/GStreamer/gstreamer-0.10.21/docs/gst' > make[4]: *** [install-am] Error 2 > make[4]: Leaving directory > `/root/Desktop/GStreamer/gstreamer-0.10.21/docs/gst' > make[3]: *** [install] Error 2 > make[3]: Leaving directory > `/root/Desktop/GStreamer/gstreamer-0.10.21/docs/gst' > make[2]: *** [install-recursive] Error 1 > make[2]: Leaving directory > `/root/Desktop/GStreamer/gstreamer-0.10.21/docs' > make[1]: *** [install] Error 2 > make[1]: Leaving directory `/root/Desktop/GStreamer/gstreamer-010.21/docs' > make: *** [install-recursive] Error 1 > > Can anybody help me to resolve this problem? > > Thank you in advance. > > > Best Regards, > Leon Liang > > > > ------------------------------------------------------------------------ > ??????????????????????????????????????5000???????? > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From felipe.contreras at gmail.com Thu Jan 15 14:06:47 2009 From: felipe.contreras at gmail.com (Felipe Contreras) Date: Thu, 15 Jan 2009 15:06:47 +0200 Subject: [gst-devel] libmpg123 decoder plugin In-Reply-To: <20090115123521.GD14461@burning-bastards.com> References: <20090114154116.GZ14461@burning-bastards.com> <94a0d4530901142147w600c113fu1b5d2cc5844b4929@mail.gmail.com> <20090115123521.GD14461@burning-bastards.com> Message-ID: <94a0d4530901150506i3cd7ffb7ydb8d829e20966d66@mail.gmail.com> On Thu, Jan 15, 2009 at 2:35 PM, Roland Poppenreiter wrote: > * On Thu, 15 Jan 2009 at 7:47 +0200, Felipe Contreras wrote: >> On Wed, Jan 14, 2009 at 5:41 PM, Roland Poppenreiter wrote: >> > Hello everybody, >> > >> > I hope this is the right mailinglist for this anouncement. I wrote >> > a mp3 decoder plugin with the mpg123 library. One reason is that >> > mpg123 has some processor optimization (SSE, MMX, 3DNow, ...) in it >> > and should be faster than the mad decoder. I used the mad decoder >> > plugin (gstmad.c) as template for my plugin - so they are quite >> > similar. I wanted to benchmark both decoders and compare the results >> > but I ran into several problems and I got a few questions: >> > >> > - sometimes i get a short white noise sound when i start playing an >> > mp3 file with my mpg123 decoder plugin. >> > - short white noise sound appears after seeking as well. >> > - is it possible that data comming from the filesrc is sometimes >> > overlapping? >> > - seeking is not always working, also not with the mad decoder. >> >> Are you using the mp3parse element? > > no, I dont even know this plugin .... > the pipeline I'm using is this: > > gst-launch filesrc location="foo.mp3" ! mpg123 ! audioconvert ! > osssink It's on the mpegaudio parse plug-in. It would do the proper frame seeking so your decoder element doesn't have to. >> Is that mpg123 library new? If I recall correctly the last time I >> tried it wasn't there, I even suggested them to create a library, but >> they didn't look too interested. > > the libmpg123 is not new, its the api from the mpg123 mp3-player > (http://www.mpg123.de/api/) and I already used it for my mpg123 > gstreamer plugin. Yeah, I know, but it was a standalone player (no library) before. Anyway, it's good that they've split it. >> Another option is to provide an OpenMAX IL API for the library. In >> that case the gst-openmax wrapper would be able to use the decoder >> transparently. > > I dont know this library, I will take a look at it, but for the > moment I have to solve my problems with the mpg123 plugin. But that's the point, if you provide the mp123 library with OpenMAX IL interface, then you don't need to write your decoder element. -- Felipe Contreras From ensonic at hora-obscura.de Thu Jan 15 13:03:33 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Thu, 15 Jan 2009 14:03:33 +0200 Subject: [gst-devel] plugin working with realimentation In-Reply-To: <1231931318.5877.742.camel@viladecans> References: <1231931318.5877.742.camel@viladecans> Message-ID: <496F2615.8010904@hora-obscura.de> hi, Antonio Marqu?s schrieb: > Hello developers, > > I am interesting to design a plugin which has as inputs N-1 number of > pads (some images from a filesrc for example) and another input which > would be the result of the plugin's output, so I would like to implement > a kind of realimentation in that plugin. > honestly, I did not understood what you want to do. Also "realimentation" seem to be a strange word (seems to be a medical term). If you want to have a history of the last N buffers you need to implement a ringbuffer that would keep a reference of them. Stefan > In that plugin I would like to have all the N input buffers in the > process function. So I have thought in using the collect pad method for > all the pads. The problem is that, for the first buffer I would have not > the buffer of the pad N (because it would be the output of the buffer), > so the pipeline would not be set on PLAYING state never. Does anyone > know how to solve this problem? The main purpose of the plugin is, to > make an initial computation for the first frame taking into account the > N-1 images and for the other frames taking also into account the > previous output, which would be the N pad. > > If anyone knows a plugin with similar function it would be great for me > to look at its source code. > > Thanks in advance. > > From thiagoss at embedded.ufcg.edu.br Thu Jan 15 15:30:33 2009 From: thiagoss at embedded.ufcg.edu.br (thiagoss) Date: Thu, 15 Jan 2009 12:30:33 -0200 Subject: [gst-devel] Building failing for qtwrapper Message-ID: Qtwraper building is failing (cvs head), I don't know if this is a misconfiguration in my system or in Makefile, so I'm sending this mail. The error output is: Making all in qtwrapper cd ../.. && /bin/sh /Users/thiagoss/Work/GStreamer/head/gst-plugins-bad/missing --run automake-1.10 --gnu sys/qtwrapper/Makefile configure.ac:111: warning: AC_CACHE_VAL(g_support_dowhile_macros, ...): suspicious cache-id, must contain _cv_ to be cached ../../lib/autoconf/general.m4:1974: AC_CACHE_VAL is expanded from... ../../lib/autoconf/general.m4:1994: AC_CACHE_CHECK is expanded from... common/m4/gst-dowhile.m4:8: AG_GST_CHECK_DOWHILE_MACROS is expanded from... configure.ac:111: the top level configure.ac:143: warning: AC_CACHE_VAL(GST_FIONREAD_IN_SYS_IOCTL, ...): suspicious cache-id, must contain _cv_ to be cached m4/gst-fionread.m4:1: GST_CHECK_FIONREAD is expanded from... configure.ac:143: the top level configure.ac:143: warning: AC_CACHE_VAL(GST_FIONREAD_IN_SYS_FILIO, ...): suspicious cache-id, must contain _cv_ to be cached cd ../.. && /bin/sh ./config.status sys/qtwrapper/Makefile depfiles config.status: creating sys/qtwrapper/Makefile config.status: executing depfiles commands /bin/sh ../../libtool --tag=CC --tag=disable-static --mode=compile gcc -DHAVE_CONFIG_H -I. -I../.. -I../../gst-libs -I../../gst-libs -D_REENTRANT -I/usr/local/include/gstreamer-0.10 -I/opt/local/include/glib-2.0 -I/opt/local/lib/glib-2.0/include -I/opt/local/include -I/opt/local/include/libxml2 -Wall -Wdeclaration-after-statement -Werror -fno-strict-aliasing -g -D_REENTRANT -I/usr/local/include/gstreamer-0.10 -I/opt/local/include/glib-2.0 -I/opt/local/lib/glib-2.0/include -I/opt/local/include -I/opt/local/include/libxml2 -g -O2 -MT libgstqtwrapper_la-qtwrapper.lo -MD -MP -MF .deps/libgstqtwrapper_la-qtwrapper.Tpo -c -o libgstqtwrapper_la-qtwrapper.lo `test -f 'qtwrapper.c' || echo './'`qtwrapper.c libtool: compile: gcc -DHAVE_CONFIG_H -I. -I../.. -I../../gst-libs -I../../gst-libs -D_REENTRANT -I/usr/local/include/gstreamer-0.10 -I/opt/local/include/glib-2.0 -I/opt/local/lib/glib-2.0/include -I/opt/local/include -I/opt/local/include/libxml2 -Wall -Wdeclaration-after-statement -Werror -fno-strict-aliasing -g -D_REENTRANT -I/usr/local/include/gstreamer-0.10 -I/opt/local/include/glib-2.0 -I/opt/local/lib/glib-2.0/include -I/opt/local/include -I/opt/local/include/libxml2 -g -O2 -MT libgstqtwrapper_la-qtwrapper.lo -MD -MP -MF .deps/libgstqtwrapper_la-qtwrapper.Tpo -c qtwrapper.c -fno-common -DPIC -o .libs/libgstqtwrapper_la-qtwrapper.o In file included from qtwrapper.c:49: qtwrapper.h:53:20: error: Movies.h: No such file or directory qtwrapper.c: In function 'plugin_init': qtwrapper.c:58: error: 'OSErr' undeclared (first use in this function) qtwrapper.c:58: error: (Each undeclared identifier is reported only once qtwrapper.c:58: error: for each function it appears in.) qtwrapper.c:58: error: syntax error before 'status' qtwrapper.c:69: error: 'status' undeclared (first use in this function) cc1: warnings being treated as errors qtwrapper.c:69: warning: implicit declaration of function 'EnterMovies' make[3]: *** [libgstqtwrapper_la-qtwrapper.lo] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 How to fix this? -- Thiago Sousa Santos Embedded Systems and Pervasive Computing Lab (Embedded) Center of Electrical Engineering and Informatics (CEEI) Federal University of Campina Grande (UFCG) -------------- next part -------------- An HTML attachment was scrubbed... URL: From dstaples at friendmts.co.uk Thu Jan 15 15:38:19 2009 From: dstaples at friendmts.co.uk (Darren Staples) Date: Thu, 15 Jan 2009 14:38:19 +0000 Subject: [gst-devel] Uridecodebin signals Message-ID: Having got a pipeline with decodebin working (programmatically) I?m now trying to do the same with uridecodebin. I?m keeping it as simple as possible and just trying to link uridecodebin ->ffmpegcolorspace -> ximagesink. I?ve read through the documentation and can see that I?m going to have to use the various signals to dynamically link uridecodebin to the next element in the pipeline. But I?m unclear as to which of the signals to use and the sequence they are triggered in. I can do this using gst-launch but this takes care of all the signal handling for me, and I want to be able to do the same with a C program. Could someone give me a brief overview of which signals I probably should use and the order they would normally occur in please ? Many thanks for all the help so far ! Darren *** FRIEND MTS LIMITED EMAIL CONFIDENTIALITY *** This email message and any attachments may contain information which is confidential or privileged and is intended for the sole use of the person to whom it is addressed. If you are not the intended recipient, be aware that any disclosures, copying, distribution or use of the contents is prohibited. If you have received this email message in error, please notify our office by telephone (+44 (0)121 633 2000) or email (admin at friendmts.co.uk) immediately. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: From wim.taymans at gmail.com Thu Jan 15 16:24:26 2009 From: wim.taymans at gmail.com (Wim Taymans) Date: Thu, 15 Jan 2009 16:24:26 +0100 Subject: [gst-devel] Uridecodebin signals In-Reply-To: References: Message-ID: <1232033066.19973.98.camel@metal> On Thu, 2009-01-15 at 14:38 +0000, Darren Staples wrote: > Having got a pipeline with decodebin working (programmatically) I?m > now trying to do the same with uridecodebin. > I?m keeping it as simple as possible and just trying to link > uridecodebin ->ffmpegcolorspace -> ximagesink. > > I?ve read through the documentation and can see that I?m going to have > to use the various signals to dynamically link uridecodebin to the > next element in the pipeline. > But I?m unclear as to which of the signals to use and the sequence > they are triggered in. > > I can do this using gst-launch but this takes care of all the signal > handling for me, and I want to be able to do the same with a C > program. > > Could someone give me a brief overview of which signals I probably > should use and the order they would normally occur in please ? You only need to connect to the pad-added signal to be notified of the dynamic pads. Wim > > Many thanks for all the help so far ! > > Darren > > > > > *** FRIEND MTS LIMITED EMAIL CONFIDENTIALITY *** > > This email message and any attachments may contain information which > is confidential or privileged and is intended for the sole use of the > person to whom it is addressed. If you are not the intended > recipient, be aware that any disclosures, copying, distribution or use > of the contents is prohibited. If you have received this email > message in error, please notify our office by telephone (+44 (0)121 > 633 2000) or email (admin at friendmts.co.uk) immediately. Thank you. > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From r_poppi at gmx.at Thu Jan 15 17:44:47 2009 From: r_poppi at gmx.at (Roland Poppenreiter) Date: Thu, 15 Jan 2009 17:44:47 +0100 Subject: [gst-devel] libmpg123 decoder plugin In-Reply-To: <94a0d4530901150506i3cd7ffb7ydb8d829e20966d66@mail.gmail.com> References: <20090114154116.GZ14461@burning-bastards.com> <94a0d4530901142147w600c113fu1b5d2cc5844b4929@mail.gmail.com> <20090115123521.GD14461@burning-bastards.com> <94a0d4530901150506i3cd7ffb7ydb8d829e20966d66@mail.gmail.com> Message-ID: <20090115164447.GF14461@burning-bastards.com> * On Thu, 15 Jan 2009 at 15:06 +0200, Felipe Contreras wrote: > On Thu, Jan 15, 2009 at 2:35 PM, Roland Poppenreiter wrote: > > * On Thu, 15 Jan 2009 at 7:47 +0200, Felipe Contreras wrote: > >> On Wed, Jan 14, 2009 at 5:41 PM, Roland Poppenreiter wrote: > >> > Hello everybody, > >> > > >> > I hope this is the right mailinglist for this anouncement. I wrote > >> > a mp3 decoder plugin with the mpg123 library. One reason is that > >> > mpg123 has some processor optimization (SSE, MMX, 3DNow, ...) in it > >> > and should be faster than the mad decoder. I used the mad decoder > >> > plugin (gstmad.c) as template for my plugin - so they are quite > >> > similar. I wanted to benchmark both decoders and compare the results > >> > but I ran into several problems and I got a few questions: > >> > > >> > - sometimes i get a short white noise sound when i start playing an > >> > mp3 file with my mpg123 decoder plugin. > >> > - short white noise sound appears after seeking as well. > >> > - is it possible that data comming from the filesrc is sometimes > >> > overlapping? > >> > - seeking is not always working, also not with the mad decoder. > >> > >> Are you using the mp3parse element? > > > > no, I dont even know this plugin .... > > the pipeline I'm using is this: > > > > gst-launch filesrc location="foo.mp3" ! mpg123 ! audioconvert ! > > osssink > > It's on the mpegaudio parse plug-in. It would do the proper frame > seeking so your decoder element doesn't have to. > > oh, thanks for the information, but anyway I solved my seeking problems without using another plugin. Also the problems with the white noise disappeared - the reason was that I initialized my GST_BUFFER (=outbuffer) with a wrong size. > >> Is that mpg123 library new? If I recall correctly the last time I > >> tried it wasn't there, I even suggested them to create a library, but > >> they didn't look too interested. > > > > the libmpg123 is not new, its the api from the mpg123 mp3-player > > (http://www.mpg123.de/api/) and I already used it for my mpg123 > > gstreamer plugin. > > Yeah, I know, but it was a standalone player (no library) before. > Anyway, it's good that they've split it. Yep, the library is good and well documented so I can recommended for everybody who wants to write an mp3 player. > >> Another option is to provide an OpenMAX IL API for the library. In > >> that case the gst-openmax wrapper would be able to use the decoder > >> transparently. > > > > I dont know this library, I will take a look at it, but for the > > moment I have to solve my problems with the mpg123 plugin. > > But that's the point, if you provide the mp123 library with OpenMAX IL > interface, then you don't need to write your decoder element. Sorry I dont get it, why should I use the mpg123 library with OpenMAX interface if I already have a working implementation of the mpg123 decoder plugin. > -- > Felipe Contreras > So now that my plugin seems to work correctly and produce a nice quality sound, even faster than the MAD decoder plugin, I would like to know if anybody want to try it? -- Roland Poppenreiter From tristan at sat.qc.ca Thu Jan 15 22:44:03 2009 From: tristan at sat.qc.ca (Tristan Matthews) Date: Thu, 15 Jan 2009 16:44:03 -0500 Subject: [gst-devel] jitter reporting from gstrtpsession Message-ID: <496FAE23.3020205@sat.qc.ca> Hi, I'm trying to find out how much jitter I have for an rtpsession. I'm using gstrtpbin. The code below outputs values but they look pretty suspect. I have this code in a callback that is invoked every 2 seconds. // session_ is a given RtpSession GValueArray *arr; GValue *val; guint i; g_object_get (session_, "sources", &arr, NULL); for (i = 0; i < arr->n_values; i++) { GObject *source; RTPSourceStats *stats; val = g_value_array_get_nth (arr, i); source = (GObject*)(g_value_get_object (val)); g_object_get(source, "stats", &stats, NULL); g_print("JITTER IS %u\n", stats->jitter); } g_value_array_free(arr); Any tips would be appreciated. I suspect that I am not getting the stats correctly. -Tristan -- Tristan Matthews Soci?t? des arts technologiques [SAT] email: tristan at sat.qc.ca web: http://www.music.mcgill.ca/~tmatthews From vaisakh.n at nestgroup.net Fri Jan 16 05:21:03 2009 From: vaisakh.n at nestgroup.net (vaisakh.n) Date: Fri, 16 Jan 2009 09:51:03 +0530 Subject: [gst-devel] Gstreamer-video. Message-ID: <49700B2F.3060608@nestgroup.net> hi, I was trying build an application to play " mpg " file so that in program i created elements are ... filesrc location=f1.mpg , mpegdemux, mpeg2dec , xvimagesink are the elements i created .. by using gst_element_factory_make(); and finaly i was able to add the dynamic pads too and i saw video but i was not able hear sound from. that.. i tried a lot to link mad ,audioconvert ,osssink using queue and something like but failed what can i do help me.... advance thanks Vaisakh.N ***** Confidentiality Statement/Disclaimer ***** This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmcouat at smartt.com Fri Jan 16 06:17:56 2009 From: rmcouat at smartt.com (Ron McOuat) Date: Thu, 15 Jan 2009 21:17:56 -0800 Subject: [gst-devel] Linux distro preferred for gstreamer? Message-ID: <49701884.5010807@smartt.com> I would like to ask a question about which Linux distro would be best for hosting a gstreamer application bundle I am working on. I hope I am not starting a distro war by asking this question, :-) just looking for some practical advice. I searched for this in the archives and didn't find much. The system I am building is a server in a corporate environment, not exposed to the public internet which is an improved version of systems that already exist but don't use gstreamer. Facts as applicable to me: 1) I am using very recent gstreamer code (stable + patches), some I did a small amount of work on which is in CVS head but not released yet so I build from source. 2) I am using Fedora 10 on my dev machine and no issues there. 3) I deploy to Centos 5.2 which is where I start to have problems. Gstreamer won't even configure because flex is too old. I pulled a Fedora 10 build tree over and I could see from trying to run that, even if I solved the flex problem I would have glib version problems which is a major base dependency of the system. This indicates it will require major work to get the latest gstreamer to run. I hear rumors of a 5.3 release which will bring in a newer set of versions for dependent libraries which may work but that is future stuff. 4) Since a lot of the field machines are running even older (2005) software than Centos 5.2 I am ready to do a mass cold metal install across them in the next few months. 5) I have dabbled with Ubuntu 8.04 and 8.10 and the latest Ubuntu is pretty much on par with Fedora wrt revision levels. If I get everything to run on 8.04 then I have an LTS release for the field which should be good for patches for quite a while. Switching to Ubuntu isn't a big problem for me other than finger memory on how to do admin things. I have used UNIX of one form or another since 1982 but the Redhat derivatives certainly feel more comfortable for me with 10 years of use. In conclusion, I find the version gap between bleeding edge and stable is wider for Redhat derived systems vs Ubuntu. This also sometimes leads to driver support issues when new hardware is purchased. I am reluctant to put Fedora into production because of the high level of update load and short patch support lifetime, so at this point I am leaning towards using Ubuntu. Any comments or sage advice from gstreamer developers? Just looking for confirmation of my conclusion or did I miss something in my evaluation? Thanks for your time, this may seem off topic but my main application dependency is gstreamer. Ron From deepakty at tataelxsi.co.in Fri Jan 16 07:21:32 2009 From: deepakty at tataelxsi.co.in (Deepak T.Y) Date: Fri, 16 Jan 2009 11:51:32 +0530 Subject: [gst-devel] Time taken by gst_pad_push function Message-ID: <9D5E1752379A43408015F7FE984661150112FDAB@CHNEXVS01.VSNLXCHANGE.COM> Hi all, I am trying to optimize my Gstreamer based application. I have a gstreamer pulgin which pushs around 35-40k data to next element in the pipeline. The following are my observations: 1. Time taken across gst_pad_push() function is around 50,000 usec. i.e gettimeofday(&t1,NULL); gst_pad_push(pad,buf); gettimeofday(&t2,NULL); Why gst_pad_push() function is taking so much time? 2. The time taken between start of gst_pad_push() and start of the chain function of next element in the pipeline is around 4,000 usec. Compared to time taken by gst_pad_push() to return to the calling fucnction, 4000 usec is less. Please can anyone help me in improving the timing of gst_pad_push()? Thanking you, Deepak T Y This message (including any attachment) is confidential and may be legally privileged. Access to this message by anyone other than the intended recipient(s) listed above is unauthorized. If you are not the intended recipient you are hereby notified that any disclosure, copying, or distribution of the message, or any action taken or omission of action by you in reliance upon it, is prohibited and may be unlawful. Please immediately notify the sender by reply e-mail and permanently delete all copies of the message if you have received this message in error. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ved.kpl at gmail.com Fri Jan 16 08:11:54 2009 From: ved.kpl at gmail.com (ved kpl) Date: Fri, 16 Jan 2009 12:41:54 +0530 Subject: [gst-devel] Time taken by gst_pad_push function In-Reply-To: <9D5E1752379A43408015F7FE984661150112FDAB@CHNEXVS01.VSNLXCHANGE.COM> References: <9D5E1752379A43408015F7FE984661150112FDAB@CHNEXVS01.VSNLXCHANGE.COM> Message-ID: <7496c23f0901152311j4d2b73e3ib440aefa12bd2b3c@mail.gmail.com> Hi, That depends on the pipeline you are using. The gst_pad_push() by the element COULD BE chained right upto downmost element, resulting in gst_pad_push() by the element eventually calling _chain() of sink element (with few elements in between). But, why do you want to profile gst_pad_push()? Ved On Fri, Jan 16, 2009 at 11:51 AM, Deepak T.Y wrote: > Hi all, > I am trying to optimize my Gstreamer based application. I have a > gstreamer pulgin > which pushs around 35-40k data to next element in the pipeline. The > following are my > observations: > > 1. Time taken across gst_pad_push() function is around 50,000 usec. i.e > gettimeofday(&t1,NULL); > gst_pad_push(pad,buf); > gettimeofday(&t2,NULL); > Why gst_pad_push() function is taking so much time? > > 2. The time taken between start of gst_pad_push() and start of the chain > function of next > element in the pipeline is around 4,000 usec. Compared to time taken by > gst_pad_push() > to return to the calling fucnction, 4000 usec is less. > > Please can anyone help me in improving the timing of gst_pad_push()? > > > Thanking you, > Deepak T Y > > > > > > > > This message (including any attachment) is confidential and may be legally > privileged. Access to this message by anyone other than the intended > recipient(s) listed above is unauthorized. If you are not the intended > recipient you are hereby notified that any disclosure, copying, or > distribution of the message, or any action taken or omission of action by > you in reliance upon it, is prohibited and may be unlawful. Please > immediately notify the sender by reply e-mail and permanently delete all > copies of the message if you have received this message in error. > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From dstaples at friendmts.co.uk Fri Jan 16 09:07:56 2009 From: dstaples at friendmts.co.uk (Darren Staples) Date: Fri, 16 Jan 2009 08:07:56 -0000 Subject: [gst-devel] Uridecodebin signals Message-ID: But when I gst-inspect uridecodebin the only signals I see are OEunknown-type?, OEautoplug-continue?, OEautoplug-factories?, OEautoplug-select? and OEdrained?. No OEpad-added? signal for uridecodebin... Or is it somehow OEinherited? from decodebin ? Or am I missing the obvious ? ;) Regards, Darren *** FRIEND MTS LIMITED EMAIL CONFIDENTIALITY *** This email message and any attachments may contain information which is confidential or privileged and is intended for the sole use of the person to whom it is addressed. If you are not the intended recipient, be aware that any disclosures, copying, distribution or use of the contents is prohibited. If you have received this email message in error, please notify our office by telephone (+44 (0)121 633 2000) or email (admin at friendmts.co.uk) immediately. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: From fthiery at gmail.com Fri Jan 16 10:16:22 2009 From: fthiery at gmail.com (Florent) Date: Fri, 16 Jan 2009 10:16:22 +0100 Subject: [gst-devel] Linux distro preferred for gstreamer? In-Reply-To: <49701884.5010807@smartt.com> References: <49701884.5010807@smartt.com> Message-ID: <1efe3a6e0901160116l717ce06ey2fbb5970d5345a7e@mail.gmail.com> Hi, I've been having a similar problematic, and we chose to stick to Ubuntu. Depending on your stable gst version requirement: * 8.04 will offer you LTS 0.10.18 gstreamer packages distribution * 8.10 will offer you 0.10.21 gstreamer packages distribution So far, we're very happy with Ubuntu hardware support and overall gstreamer stability; but i aknowledge the lack of a lightweight gstreamer-oriented distribution for dedicated multimedia solutions deployment. It would be great to start such a project ! Florent From deepakty at tataelxsi.co.in Fri Jan 16 10:28:30 2009 From: deepakty at tataelxsi.co.in (Deepak T.Y) Date: Fri, 16 Jan 2009 14:58:30 +0530 Subject: [gst-devel] Time taken by gst_pad_push function References: <9D5E1752379A43408015F7FE984661150112FDAB@CHNEXVS01.VSNLXCHANGE.COM> <7496c23f0901152311j4d2b73e3ib440aefa12bd2b3c@mail.gmail.com> Message-ID: <9D5E1752379A43408015F7FE984661150112FDAE@CHNEXVS01.VSNLXCHANGE.COM> Hi Ved, Thanks for your reply. The element(lets say X) where I am calling gst_pad_push(), pumps data to next element. Untill gst_pad_push returns back, element X doesn't start processing for the next push. Effectively my gstreamer pipeline has become sequential. 1. Currently I dont have any "queue" element in my pipeline. Will it help If I introduce "queue" between every element in my pipeline? 2. I guess my pipeline is a downstream one. Will it the performance of my application improve if I make it upstream? With Regards, Deepak T Y -----Original Message----- From: ved kpl [mailto:ved.kpl at gmail.com] Sent: Fri 1/16/2009 12:41 PM To: Discussion of the development of GStreamer Subject: Re: [gst-devel] Time taken by gst_pad_push function Hi, That depends on the pipeline you are using. The gst_pad_push() by the element COULD BE chained right upto downmost element, resulting in gst_pad_push() by the element eventually calling _chain() of sink element (with few elements in between). But, why do you want to profile gst_pad_push()? Ved On Fri, Jan 16, 2009 at 11:51 AM, Deepak T.Y wrote: > Hi all, > I am trying to optimize my Gstreamer based application. I have a > gstreamer pulgin > which pushs around 35-40k data to next element in the pipeline. The > following are my > observations: > > 1. Time taken across gst_pad_push() function is around 50,000 usec. i.e > gettimeofday(&t1,NULL); > gst_pad_push(pad,buf); > gettimeofday(&t2,NULL); > Why gst_pad_push() function is taking so much time? > > 2. The time taken between start of gst_pad_push() and start of the chain > function of next > element in the pipeline is around 4,000 usec. Compared to time taken by > gst_pad_push() > to return to the calling fucnction, 4000 usec is less. > > Please can anyone help me in improving the timing of gst_pad_push()? > > > Thanking you, > Deepak T Y > > > > > > > > This message (including any attachment) is confidential and may be legally > privileged. Access to this message by anyone other than the intended > recipient(s) listed above is unauthorized. If you are not the intended > recipient you are hereby notified that any disclosure, copying, or > distribution of the message, or any action taken or omission of action by > you in reliance upon it, is prohibited and may be unlawful. Please > immediately notify the sender by reply e-mail and permanently delete all > copies of the message if you have received this message in error. > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ------------------------------------------------------------------------------ This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel This message (including any attachment) is confidential and may be legally privileged. Access to this message by anyone other than the intended recipient(s) listed above is unauthorized. If you are not the intended recipient you are hereby notified that any disclosure, copying, or distribution of the message, or any action taken or omission of action by you in reliance upon it, is prohibited and may be unlawful. Please immediately notify the sender by reply e-mail and permanently delete all copies of the message if you have received this message in error. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 4288 bytes Desc: not available URL: From ved.kpl at gmail.com Fri Jan 16 11:16:36 2009 From: ved.kpl at gmail.com (ved kpl) Date: Fri, 16 Jan 2009 15:46:36 +0530 Subject: [gst-devel] Time taken by gst_pad_push function In-Reply-To: <9D5E1752379A43408015F7FE984661150112FDAE@CHNEXVS01.VSNLXCHANGE.COM> References: <9D5E1752379A43408015F7FE984661150112FDAB@CHNEXVS01.VSNLXCHANGE.COM> <7496c23f0901152311j4d2b73e3ib440aefa12bd2b3c@mail.gmail.com> <9D5E1752379A43408015F7FE984661150112FDAE@CHNEXVS01.VSNLXCHANGE.COM> Message-ID: <7496c23f0901160216v3e5626cah68a89a884924dace@mail.gmail.com> On Fri, Jan 16, 2009 at 2:58 PM, Deepak T.Y wrote: > Hi Ved, > Thanks for your reply. > The element(lets say X) where I am calling gst_pad_push(), pumps data > to next element. Untill gst_pad_push returns back, element X doesn't > start processing for the next push. Effectively my gstreamer pipeline > has become sequential. > > 1. Currently I dont have any "queue" element in my pipeline. > Will it help If I introduce "queue" between every element in my pipeline? You can use queue if you want to decouple two elements. Queue starts a different thread to push data downstream. It entirely depends on your pipeline design. Queue accepts data form the thread above it self and pushes the data downstream in a different thread. > > 2. I guess my pipeline is a downstream one. Will it the performance of > my application improve if I make it upstream? Sorry, I havent understood the question. What do you mean by making the pipeline upstream? > > With Regards, > Deepak T Y > > > -----Original Message----- > From: ved kpl [mailto:ved.kpl at gmail.com] > Sent: Fri 1/16/2009 12:41 PM > To: Discussion of the development of GStreamer > Subject: Re: [gst-devel] Time taken by gst_pad_push function > > Hi, > > That depends on the pipeline you are using. The gst_pad_push() by the > element COULD BE chained right upto downmost element, resulting in > gst_pad_push() by the element eventually calling _chain() of sink > element (with few elements in between). > > But, why do you want to profile gst_pad_push()? > > Ved > > On Fri, Jan 16, 2009 at 11:51 AM, Deepak T.Y wrote: >> Hi all, >> I am trying to optimize my Gstreamer based application. I have a >> gstreamer pulgin >> which pushs around 35-40k data to next element in the pipeline. The >> following are my >> observations: >> >> 1. Time taken across gst_pad_push() function is around 50,000 usec. i.e >> gettimeofday(&t1,NULL); >> gst_pad_push(pad,buf); >> gettimeofday(&t2,NULL); >> Why gst_pad_push() function is taking so much time? >> >> 2. The time taken between start of gst_pad_push() and start of the chain >> function of next >> element in the pipeline is around 4,000 usec. Compared to time taken by >> gst_pad_push() >> to return to the calling fucnction, 4000 usec is less. >> >> Please can anyone help me in improving the timing of gst_pad_push()? >> >> >> Thanking you, >> Deepak T Y >> >> >> >> >> >> >> >> This message (including any attachment) is confidential and may be legally >> privileged. Access to this message by anyone other than the intended >> recipient(s) listed above is unauthorized. If you are not the intended >> recipient you are hereby notified that any disclosure, copying, or >> distribution of the message, or any action taken or omission of action by >> you in reliance upon it, is prohibited and may be unlawful. Please >> immediately notify the sender by reply e-mail and permanently delete all >> copies of the message if you have received this message in error. >> >> ------------------------------------------------------------------------------ >> This SF.net email is sponsored by: >> SourcForge Community >> SourceForge wants to tell your story. >> http://p.sf.net/sfu/sf-spreadtheword >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > This message (including any attachment) is confidential and may be legally privileged. Access to this message by anyone other than the intended recipient(s) listed above is unauthorized. If you are not the intended recipient you are hereby notified that any disclosure, copying, or distribution of the message, or any action taken or omission of action by you in reliance upon it, is prohibited and may be unlawful. Please immediately notify the sender by reply e-mail and permanently delete all copies of the message if you have received this message in error. > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From sumanth.v at allaboutif.com Fri Jan 16 11:44:43 2009 From: sumanth.v at allaboutif.com (Sumanth V) Date: Fri, 16 Jan 2009 16:14:43 +0530 Subject: [gst-devel] DVD playback using dvdnavsrc Message-ID: <83b2c1480901160244i1c1a5bccg27180563cdabe521@mail.gmail.com> Hi all, I am writing a DVD Play back application using dvdnavsrc plugin. It is working fine except the MENU highlight. Option in the MENU screen are not getting highlighted. but it is able to display the subtitles. How to fix this issue in dvdnavsrc plugin?? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: From vaisakh.n at nestgroup.net Fri Jan 16 12:22:53 2009 From: vaisakh.n at nestgroup.net (vaisakh.n) Date: Fri, 16 Jan 2009 16:52:53 +0530 Subject: [gst-devel] gtreamer+plugin Message-ID: <49706E0D.9020800@nestgroup.net> hi, Can i add a new plugin( which is completly new not wrriten under open source category ) to gstreamer... how can i do that?? help me is there any specific API 's in gstreamer??? ?? ***** Confidentiality Statement/Disclaimer ***** This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. From thaytan at noraisin.net Fri Jan 16 14:23:53 2009 From: thaytan at noraisin.net (Jan Schmidt) Date: Fri, 16 Jan 2009 13:23:53 +0000 Subject: [gst-devel] DVD playback using dvdnavsrc In-Reply-To: <83b2c1480901160244i1c1a5bccg27180563cdabe521@mail.gmail.com> References: <83b2c1480901160244i1c1a5bccg27180563cdabe521@mail.gmail.com> Message-ID: <1232112233.12979.1.camel@fancy-ubuntu> On Fri, 2009-01-16 at 16:14 +0530, Sumanth V wrote: > Hi all, > I am writing a DVD Play back application using dvdnavsrc > plugin. It is working fine except the MENU highlight. Option in the > MENU screen are not getting highlighted. but it is able to display the > subtitles. How to fix this issue in dvdnavsrc plugin?? > dvdnavsrc doesn't work particularly well, and I have no plans to improve it. I'm focussing my development on the ResinDVD components in gst-plugins-bad, which already work better for most DVDs. Try using the latest rsndvdbin from gst-plugins-bad-0.10.9.3 (which will be gst-plugins-bad 0.10.10 on Monday), instead of dvdnavsrc. Regards, Jan. -- Jan Schmidt From ensonic at hora-obscura.de Fri Jan 16 14:26:15 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Fri, 16 Jan 2009 15:26:15 +0200 Subject: [gst-devel] audio and video plugin base classes Message-ID: <49708AF7.40302@hora-obscura.de> hi, I would like to see some more features covered in audio base classes. Its better to have it in baseclasses than just implemented in a similar way in various elements, as in the baseclass case apps can rely on it :) http://www.buzztard.org/index.php/Audio_base_classes if this is interesting for video too, I could move it to the gst-wiki and we widen the scope. Stefan From sumanth.v at allaboutif.com Fri Jan 16 14:34:50 2009 From: sumanth.v at allaboutif.com (Sumanth V) Date: Fri, 16 Jan 2009 19:04:50 +0530 Subject: [gst-devel] DVD playback using dvdnavsrc In-Reply-To: <1232112233.12979.1.camel@fancy-ubuntu> References: <83b2c1480901160244i1c1a5bccg27180563cdabe521@mail.gmail.com> <1232112233.12979.1.camel@fancy-ubuntu> Message-ID: <83b2c1480901160534m28bd4256i79a799a6c4a33330@mail.gmail.com> Thanks.. On Fri, Jan 16, 2009 at 6:53 PM, Jan Schmidt wrote: > On Fri, 2009-01-16 at 16:14 +0530, Sumanth V wrote: > > Hi all, > > I am writing a DVD Play back application using dvdnavsrc > > plugin. It is working fine except the MENU highlight. Option in the > > MENU screen are not getting highlighted. but it is able to display the > > subtitles. How to fix this issue in dvdnavsrc plugin?? > > > > dvdnavsrc doesn't work particularly well, and I have no plans to improve > it. I'm focussing my development on the ResinDVD components in > gst-plugins-bad, which already work better for most DVDs. > > Try using the latest rsndvdbin from gst-plugins-bad-0.10.9.3 (which will > be gst-plugins-bad 0.10.10 on Monday), instead of dvdnavsrc. > > Regards, > Jan. > -- > Jan Schmidt > > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From thiagoss at embedded.ufcg.edu.br Fri Jan 16 12:36:55 2009 From: thiagoss at embedded.ufcg.edu.br (thiagoss) Date: Fri, 16 Jan 2009 08:36:55 -0300 Subject: [gst-devel] gtreamer+plugin In-Reply-To: <49706E0D.9020800@nestgroup.net> References: <49706E0D.9020800@nestgroup.net> Message-ID: On Fri, Jan 16, 2009 at 8:22 AM, vaisakh.n wrote: > hi, > > Can i add a new plugin( which is completly new not wrriten under open > source category ) to gstreamer... > > how can i do that?? > help me > is there any specific API 's in gstreamer??? > ?? You should read the docs in the website (which is currently down for maintenance). Read the application development manual and then the plugins writing guide. > > ***** Confidentiality Statement/Disclaimer ***** > > This message and any attachments is intended for the sole use of the > intended recipient. It may contain confidential information. Any > unauthorized use, dissemination or modification is strictly prohibited. If > you are not the intended recipient, please notify the sender immediately > then delete it from all your systems, and do not copy, use or print. > Internet communications are not secure and it is the responsibility of the > recipient to make sure that it is virus/malicious code exempt. > The company/sender cannot be responsible for any unauthorized alterations > or modifications made to the contents. If you require any form of > confirmation of the contents, please contact the company/sender. The > company/sender is not liable for any errors or omissions in the content of > this message. > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Thiago Sousa Santos Embedded Systems and Pervasive Computing Lab (Embedded) Center of Electrical Engineering and Informatics (CEEI) Federal University of Campina Grande (UFCG) -------------- next part -------------- An HTML attachment was scrubbed... URL: From t.i.m at zen.co.uk Fri Jan 16 14:40:17 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Fri, 16 Jan 2009 13:40:17 +0000 Subject: [gst-devel] gtreamer+plugin In-Reply-To: <49706E0D.9020800@nestgroup.net> References: <49706E0D.9020800@nestgroup.net> Message-ID: <1232113217.4751.9.camel@mini.centricular.net> On Fri, 2009-01-16 at 16:52 +0530, vaisakh.n wrote: > Can i add a new plugin( which is completly new not written under open > source category ) to gstreamer... That depends on what you want/need. There is nothing special about such a plugin. It works just like with any other plugin, only that you would specify "Proprietary" as license in the license field and not e.g. "LGPL". > how can i do that?? > help me > is there any specific API 's in gstreamer??? > ?? You have two options: 1. If you want to make the plugin available to all GStreamer applications, you need to make sure it gets installed into the right plugin directory. This is no different from any other plugin module. 2. If the plugin is specific to an application of yours, there is no need to install it system-wide - you can simply ship it as part of your application binary and register it as 'static plugin' just after you call gst_init(). See gst_plugin_register_static(). I'm sure you can find an example somewhere if you look for it. Cheers -Tim PS: one question mark is usually sufficient From t.i.m at zen.co.uk Fri Jan 16 15:09:36 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Fri, 16 Jan 2009 14:09:36 +0000 Subject: [gst-devel] gdkpixbufsink factory not found In-Reply-To: <20090114094939.u964aro9w0ckww4s@maggie.enixns.com> References: <20090114094939.u964aro9w0ckww4s@maggie.enixns.com> Message-ID: <1232114976.4751.20.camel@mini.centricular.net> On Wed, 2009-01-14 at 09:49 +0000, mapping at sheerman-chase.org.uk wrote: Hi, > A quick follow up, in which I have found my problem but not a > solution. Going gst-inspect gdkpixbuf shows I don't have the > gdkpixbufsink factory plugin in my version of gdkpixbuf (0.10.4). Upgrade! :) > My next step is to investigate if using gdkpixbufdec is a possibility > instead of gdkpixbufsink. No, gdkpixbufdec decodes images e.g. from PNG or JPEG to raw video. gdkpixbufsink merely encapsulates raw RGB video data in a GdkPixbuf struct. > Or possibly use a different pixel buffer (if any exist).Do I need > to have a sink element after gdkpixbufdec? If so, which one? > > Is there any logical way to indicate when the buffer is updated? > (basically to reproduce gdkpixbufsink's behaviour) gdkpixbufsink is not a complicated element - you could (licenses permitting) just copy it into your application's source tree, prefix all types and names with your own namespace (e.g. MyGdkPixbufSink etc.) and ship it as static plugin with your app. Or reimplement the bits you need, which is probably not much code either. Alternatively, you could use a capsfilter ! fakesink where capsfilter is set to the desired RGB caps, and then handle fakesink's handoff and/or preroll-handoff signals (you can post messages on the bus yourself from the callbacks if you want to). This gets you raw RGB pixel data. There's also appsink which is now stable and in the upcoming -base release, but if you don't have the latest -good, you probably don't have that either. Cheers -Tim From rmcouat at smartt.com Fri Jan 16 17:30:03 2009 From: rmcouat at smartt.com (Ron McOuat) Date: Fri, 16 Jan 2009 08:30:03 -0800 Subject: [gst-devel] Linux distro preferred for gstreamer? In-Reply-To: <1efe3a6e0901160116l717ce06ey2fbb5970d5345a7e@mail.gmail.com> References: <49701884.5010807@smartt.com> <1efe3a6e0901160116l717ce06ey2fbb5970d5345a7e@mail.gmail.com> Message-ID: <4970B60B.6090103@smartt.com> Thanks Florent for your kind advice. Waking up this AM I realized what has been bothering me about the Fedora-RHEL versions, not to say bad things about them, they work very hard to produce a good product that I have been thankful to have for many years. The realization is during development of something one tends to need the newest releases. However, no matter how long I wait, a Fedora release will not seamlessly become a RHEL version without re-install. If I have to use Ubuntu 8.10 to get the latest then I can slide that system along to the next LTS e.g. 9.10 for a hypothetical number and if development is done on that system I can lock it in for 5 years if server or 3 years if desktop. It all stems from the respective business models of the two organizations. Your comments on a lighter server based gstreamer are very relevant to me as well. I have a Ubuntu Server 8.04 machine installed which is very light but with no gstreamer used for backup. On top of that can be installed a metapackage to turn it into Ubuntu-Gnome, Kubuntu-KDE or Xubuntu-XFCE which is a full meal deal, ok Xubuntu is a 3/4 meal if some desktop features are needed. It should be possible to build a metapackage that installs the gstreramer plus dependency essentials for server use. If X is left off the machine then there will be no display function so maybe it would need an X/non-X version. Of course I may get caught by other dependencies. When I get closer to completion with what I am working on I will look into how to build a metapackage. I don't believe it amounts to much more than a list of the needed components put into a wrapper. Ron Florent wrote: > Hi, > > I've been having a similar problematic, and we chose to stick to > Ubuntu. Depending on your stable gst version requirement: > * 8.04 will offer you LTS 0.10.18 gstreamer packages distribution > * 8.10 will offer you 0.10.21 gstreamer packages distribution > > So far, we're very happy with Ubuntu hardware support and overall > gstreamer stability; but i aknowledge the lack of a lightweight > gstreamer-oriented distribution for dedicated multimedia solutions > deployment. It would be great to start such a project ! > > Florent > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From bilboed at gmail.com Fri Jan 16 23:14:33 2009 From: bilboed at gmail.com (Edward Hervey) Date: Fri, 16 Jan 2009 23:14:33 +0100 Subject: [gst-devel] Linux distro preferred for gstreamer? In-Reply-To: <49701884.5010807@smartt.com> References: <49701884.5010807@smartt.com> Message-ID: <1232144073.5032.24.camel@joder> Hi, Just thought I'd make the following information available. In order for people to test new (pre-)releases of pitivi on ubuntu, we've set up a PPA for all GStreamer packages from debian/experimental here: https://launchpad.net/~gstreamer-developers We try to update it as soon as debian/experimental packages are updated. It's as 'bleeding-edge' as you can get on ubuntu without having to do any compiling :) Edward On Thu, 2009-01-15 at 21:17 -0800, Ron McOuat wrote: > I would like to ask a question about which Linux distro would be best > for hosting a gstreamer application bundle I am working on. I hope I am > not starting a distro war by asking this question, :-) just looking for > some practical advice. I searched for this in the archives and didn't > find much. The system I am building is a server in a corporate > environment, not exposed to the public internet which is an improved > version of systems that already exist but don't use gstreamer. > > Facts as applicable to me: > > 1) I am using very recent gstreamer code (stable + patches), some I did > a small amount of work on which is in CVS head but not released yet so I > build from source. > > 2) I am using Fedora 10 on my dev machine and no issues there. > > 3) I deploy to Centos 5.2 which is where I start to have problems. > Gstreamer won't even configure because flex is too old. I pulled a > Fedora 10 build tree over and I could see from trying to run that, even > if I solved the flex problem I would have glib version problems which is > a major base dependency of the system. This indicates it will require > major work to get the latest gstreamer to run. I hear rumors of a 5.3 > release which will bring in a newer set of versions for dependent > libraries which may work but that is future stuff. > > 4) Since a lot of the field machines are running even older (2005) > software than Centos 5.2 I am ready to do a mass cold metal install > across them in the next few months. > > 5) I have dabbled with Ubuntu 8.04 and 8.10 and the latest Ubuntu is > pretty much on par with Fedora wrt revision levels. If I get everything > to run on 8.04 then I have an LTS release for the field which should be > good for patches for quite a while. Switching to Ubuntu isn't a big > problem for me other than finger memory on how to do admin things. I > have used UNIX of one form or another since 1982 but the Redhat > derivatives certainly feel more comfortable for me with 10 years of use. > > In conclusion, I find the version gap between bleeding edge and stable > is wider for Redhat derived systems vs Ubuntu. This also sometimes leads > to driver support issues when new hardware is purchased. I am reluctant > to put Fedora into production because of the high level of update load > and short patch support lifetime, so at this point I am leaning towards > using Ubuntu. > > Any comments or sage advice from gstreamer developers? Just looking for > confirmation of my conclusion or did I miss something in my evaluation? > > Thanks for your time, this may seem off topic but my main application > dependency is gstreamer. > > Ron > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From t.i.m at zen.co.uk Fri Jan 16 23:52:44 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Fri, 16 Jan 2009 22:52:44 +0000 Subject: [gst-devel] Linux distro preferred for gstreamer? In-Reply-To: <49701884.5010807@smartt.com> References: <49701884.5010807@smartt.com> Message-ID: <1232146364.17616.9.camel@mini.centricular.net> On Thu, 2009-01-15 at 21:17 -0800, Ron McOuat wrote: > I would like to ask a question about which Linux distro would be best > for hosting a gstreamer application bundle I am working on. > > (snipalot) > > Any comments or sage advice from gstreamer developers? Just looking for > confirmation of my conclusion or did I miss something in my evaluation? > > Thanks for your time, this may seem off topic but my main application > dependency is gstreamer. I'd recommend a recent debian or Ubuntu or Fedora Core (in no particular order). All of these have fairly well maintained GStreamer packages. Cheers -Tim From vacek008 at umn.edu Fri Jan 16 23:08:08 2009 From: vacek008 at umn.edu (Tom Vacek) Date: Fri, 16 Jan 2009 16:08:08 -0600 Subject: [gst-devel] msgsm plugin not recognized Message-ID: <49710548.6020101@umn.edu> A voicemail system (Asterisk) forwards messages to me in "msgsm" format. I can play them with Blastwave mplayer (see below). In a nutshell, I compiled a gstreamer msgsm plugin, and gstreamer appears to recognize it, but my messages still won't play. The following three command outputs tell the story: ====================================== xx at v1:~$ gst-inspect msgsm Plugin Details: Name: msgsm Description: msGSM encoder/decoder Filename: /usr/lib/gstreamer-0.10/libgstmsgsm.so Version: 0.12.10 License: LGPL Source module: gst-plugins-farsight Binary package: Farsight Origin URL: http://farsight.sf.net msgsmenc: MS GSM audio encoder msgsmdec: MS GSM audio decoder 2 features: +-- 2 elements ============================ xx at v1:~$ totem msg0000.WAV ** Message: don't know how to handle audio/ms-gsm, rate=(int)8000, channels=(int)1, codec_data=(buffer)4001 ** Message: Error: You do not have a decoder installed to handle this file. You might need to install the necessary plugins. gstplaybasebin.c(2307): prepare_output (): /GstPlayBin:play ** Message: Missing plugin: gstreamer|0.10|totem|MS GSM decoder|decoder-audio/ms-gsm (MS GSM decoder) ** Message: Automatic missing codec installation not supported (helper script missing) ======================================= xx at v1:~$ mplayer msg0000.WAV MPlayer 1.0rc2-CSW-Studio11 (C) 2000-2007 MPlayer Team CPU: Intel x86 CPUflags: MMX: 0 MMX2: 0 3DNow: 0 3DNow2: 0 SSE: 0 SSE2: 0 Compiled for x86 CPU with extensions: Playing msg0000.WAV. Audio file file format detected. ========================================================================== Opening audio decoder: [msgsm] native GSM/MSGSM audio decoder AUDIO: 8000 Hz, 1 ch, s16le, 13.0 kbit/10.16% (ratio: 1625->16000) Selected audio codec: [msgsm] afm: msgsm (MS GSM) ========================================================================== From msmith at xiph.org Sat Jan 17 00:00:56 2009 From: msmith at xiph.org (Michael Smith) Date: Fri, 16 Jan 2009 15:00:56 -0800 Subject: [gst-devel] msgsm plugin not recognized In-Reply-To: <49710548.6020101@umn.edu> References: <49710548.6020101@umn.edu> Message-ID: <3c1737210901161500o13bbf362ide420f2b8fe8b2d0@mail.gmail.com> On Fri, Jan 16, 2009 at 2:08 PM, Tom Vacek wrote: > A voicemail system (Asterisk) forwards messages to me in "msgsm" > format. I can play them with Blastwave mplayer (see below). In a > nutshell, I compiled a gstreamer msgsm plugin, and gstreamer appears to > recognize it, but my messages still won't play. > > The following three command outputs tell the story: Not really. "gst-inspect msgsmdec" would be the crucial missing info... Mike > > ====================================== > xx at v1:~$ gst-inspect msgsm > Plugin Details: > Name: msgsm > Description: msGSM encoder/decoder > Filename: /usr/lib/gstreamer-0.10/libgstmsgsm.so > Version: 0.12.10 > License: LGPL > Source module: gst-plugins-farsight > Binary package: Farsight > Origin URL: http://farsight.sf.net > > msgsmenc: MS GSM audio encoder > msgsmdec: MS GSM audio decoder > > 2 features: > +-- 2 elements > > ============================ > xx at v1:~$ totem msg0000.WAV > ** Message: don't know how to handle audio/ms-gsm, rate=(int)8000, > channels=(int)1, codec_data=(buffer)4001 > ** Message: Error: You do not have a decoder installed to handle this > file. You might need to install the necessary plugins. > gstplaybasebin.c(2307): prepare_output (): /GstPlayBin:play > > ** Message: Missing plugin: gstreamer|0.10|totem|MS GSM > decoder|decoder-audio/ms-gsm (MS GSM decoder) > ** Message: Automatic missing codec installation not supported (helper > script missing) > > ======================================= > xx at v1:~$ mplayer msg0000.WAV > MPlayer 1.0rc2-CSW-Studio11 (C) 2000-2007 MPlayer Team > CPU: Intel x86 > CPUflags: MMX: 0 MMX2: 0 3DNow: 0 3DNow2: 0 SSE: 0 SSE2: 0 > Compiled for x86 CPU with extensions: > > Playing msg0000.WAV. > Audio file file format detected. > ========================================================================== > Opening audio decoder: [msgsm] native GSM/MSGSM audio decoder > AUDIO: 8000 Hz, 1 ch, s16le, 13.0 kbit/10.16% (ratio: 1625->16000) > Selected audio codec: [msgsm] afm: msgsm (MS GSM) > ========================================================================== > > > > > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From vacek008 at umn.edu Sat Jan 17 00:03:51 2009 From: vacek008 at umn.edu (Tom Vacek) Date: Fri, 16 Jan 2009 17:03:51 -0600 Subject: [gst-devel] msgsm plugin not recognized In-Reply-To: <3c1737210901161500o13bbf362ide420f2b8fe8b2d0@mail.gmail.com> References: <49710548.6020101@umn.edu> <3c1737210901161500o13bbf362ide420f2b8fe8b2d0@mail.gmail.com> Message-ID: <49711257.3000006@umn.edu> Thanks for looking at this. Here it is. twvacek at v1:~/Desktop$ gst-inspect msgsmdec Factory Details: Long name: MS GSM audio decoder Class: Codec/Decoder/Audio Description: Decodes MS GSM encoded audio Author(s): Philippe Khalaf Rank: none (0) Plugin Details: Name: msgsm Description: msGSM encoder/decoder Filename: /usr/lib/gstreamer-0.10/libgstmsgsm.so Version: 0.12.10 License: LGPL Source module: gst-plugins-farsight Binary package: Farsight Origin URL: http://farsight.sf.net GObject +----GstObject +----GstElement +----GstMSGSMDec Pad Templates: SINK template: 'sink' Availability: Always Capabilities: audio/x-msgsm rate: 8000 channels: 1 SRC template: 'src' Availability: Always Capabilities: audio/x-raw-int endianness: 1234 signed: true width: 16 depth: 16 rate: 8000 channels: 1 Element Flags: no flags set Element Implementation: Has change_state() function: gst_element_change_state_func Has custom save_thyself() function: gst_element_save_thyself Has custom restore_thyself() function: gst_element_restore_thyself Element has no clocking capabilities. Element has no indexing capabilities. Element has no URI handling capabilities. Pads: SRC: 'src' Implementation: Has custom eventfunc(): gst_pad_event_default Has custom queryfunc(): gst_pad_query_default Provides query types: Has custom intconnfunc(): gst_pad_get_internal_links_default Pad Template: 'src' SINK: 'sink' Implementation: Has chainfunc(): gst_msgsmdec_chain Has custom eventfunc(): gst_pad_event_default Has custom queryfunc(): gst_pad_query_default Provides query types: Has custom intconnfunc(): gst_pad_get_internal_links_default Pad Template: 'sink' Element Properties: name : The name of the object flags: readable, writable String. Default: null Current: "msgsmdec0" Michael Smith wrote: > On Fri, Jan 16, 2009 at 2:08 PM, Tom Vacek wrote: > >> A voicemail system (Asterisk) forwards messages to me in "msgsm" >> format. I can play them with Blastwave mplayer (see below). In a >> nutshell, I compiled a gstreamer msgsm plugin, and gstreamer appears to >> recognize it, but my messages still won't play. >> >> The following three command outputs tell the story: >> > > Not really. "gst-inspect msgsmdec" would be the crucial missing info... > > Mike > > >> ====================================== >> xx at v1:~$ gst-inspect msgsm >> Plugin Details: >> Name: msgsm >> Description: msGSM encoder/decoder >> Filename: /usr/lib/gstreamer-0.10/libgstmsgsm.so >> Version: 0.12.10 >> License: LGPL >> Source module: gst-plugins-farsight >> Binary package: Farsight >> Origin URL: http://farsight.sf.net >> >> msgsmenc: MS GSM audio encoder >> msgsmdec: MS GSM audio decoder >> >> 2 features: >> +-- 2 elements >> >> ============================ >> xx at v1:~$ totem msg0000.WAV >> ** Message: don't know how to handle audio/ms-gsm, rate=(int)8000, >> channels=(int)1, codec_data=(buffer)4001 >> ** Message: Error: You do not have a decoder installed to handle this >> file. You might need to install the necessary plugins. >> gstplaybasebin.c(2307): prepare_output (): /GstPlayBin:play >> >> ** Message: Missing plugin: gstreamer|0.10|totem|MS GSM >> decoder|decoder-audio/ms-gsm (MS GSM decoder) >> ** Message: Automatic missing codec installation not supported (helper >> script missing) >> >> ======================================= >> xx at v1:~$ mplayer msg0000.WAV >> MPlayer 1.0rc2-CSW-Studio11 (C) 2000-2007 MPlayer Team >> CPU: Intel x86 >> CPUflags: MMX: 0 MMX2: 0 3DNow: 0 3DNow2: 0 SSE: 0 SSE2: 0 >> Compiled for x86 CPU with extensions: >> >> Playing msg0000.WAV. >> Audio file file format detected. >> ========================================================================== >> Opening audio decoder: [msgsm] native GSM/MSGSM audio decoder >> AUDIO: 8000 Hz, 1 ch, s16le, 13.0 kbit/10.16% (ratio: 1625->16000) >> Selected audio codec: [msgsm] afm: msgsm (MS GSM) >> ========================================================================== >> >> >> >> >> >> >> ------------------------------------------------------------------------------ >> This SF.net email is sponsored by: >> SourcForge Community >> SourceForge wants to tell your story. >> http://p.sf.net/sfu/sf-spreadtheword >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From bilboed at gmail.com Fri Jan 16 23:20:46 2009 From: bilboed at gmail.com (Edward Hervey) Date: Fri, 16 Jan 2009 23:20:46 +0100 Subject: [gst-devel] Uridecodebin signals In-Reply-To: References: Message-ID: <1232144446.5032.27.camel@joder> Hi, gst-inspect only shows the signals defined in the highest-level class. Since 'pad-added' is a signal of GstElement (of which all elements derive, including bins and pipelines) it doesn't show it... but it's there :) Edward On Fri, 2009-01-16 at 08:07 +0000, Darren Staples wrote: > But when I gst-inspect uridecodebin the only signals I see are > ?unknown-type?, ?autoplug-continue?, ?autoplug-factories?, > ?autoplug-select? and ?drained?. > No ?pad-added? signal for uridecodebin... > > Or is it somehow ?inherited? from decodebin ? Or am I missing the > obvious ? ;) > > Regards, > Darren > > > > > *** FRIEND MTS LIMITED EMAIL CONFIDENTIALITY *** > > This email message and any attachments may contain information which > is confidential or privileged and is intended for the sole use of the > person to whom it is addressed. If you are not the intended > recipient, be aware that any disclosures, copying, distribution or use > of the contents is prohibited. If you have received this email > message in error, please notify our office by telephone (+44 (0)121 > 633 2000) or email (admin at friendmts.co.uk) immediately. Thank you. > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From msmith at xiph.org Sat Jan 17 00:57:14 2009 From: msmith at xiph.org (Michael Smith) Date: Fri, 16 Jan 2009 15:57:14 -0800 Subject: [gst-devel] msgsm plugin not recognized In-Reply-To: <49711257.3000006@umn.edu> References: <49710548.6020101@umn.edu> <3c1737210901161500o13bbf362ide420f2b8fe8b2d0@mail.gmail.com> <49711257.3000006@umn.edu> Message-ID: <3c1737210901161557i2e166b10yc99018b37bd7f93d@mail.gmail.com> Tom, So, from your first email we have this:> Capabilities: > audio/x-msgsm > rate: 8000 > channels: 1 ** Message: don't know how to handle audio/ms-gsm, rate=(int)8000, channels=(int)1, codec_data=(buffer)4001 >From the gst-inspect output, we have this: Capabilities: audio/x-msgsm rate: 8000 channels: 1 This means that the msgsmdec decoder that you have accepts a type called "audio/x-msgsm". However, the demuxer outputs a type called "audio/ms-gsm". These aren't the same, so the decoder is considered incompatible, and therefore not used. You'll need to change the decoder to use "audio/ms-gsm" for this to work. Mike On Fri, Jan 16, 2009 at 3:03 PM, Tom Vacek wrote: > Thanks for looking at this. Here it is. > > twvacek at v1:~/Desktop$ gst-inspect msgsmdec > Factory Details: > Long name: MS GSM audio decoder > Class: Codec/Decoder/Audio > Description: Decodes MS GSM encoded audio > Author(s): Philippe Khalaf > Rank: none (0) > > Plugin Details: > Name: msgsm > Description: msGSM encoder/decoder > Filename: /usr/lib/gstreamer-0.10/libgstmsgsm.so > Version: 0.12.10 > License: LGPL > Source module: gst-plugins-farsight > Binary package: Farsight > Origin URL: http://farsight.sf.net > > GObject > +----GstObject > +----GstElement > +----GstMSGSMDec > > Pad Templates: > SINK template: 'sink' > Availability: Always > Capabilities: > audio/x-msgsm > rate: 8000 > channels: 1 > > SRC template: 'src' > Availability: Always > Capabilities: > audio/x-raw-int > endianness: 1234 > signed: true > width: 16 > depth: 16 > rate: 8000 > channels: 1 > > > Element Flags: > no flags set > > Element Implementation: > Has change_state() function: gst_element_change_state_func > Has custom save_thyself() function: gst_element_save_thyself > Has custom restore_thyself() function: gst_element_restore_thyself > > Element has no clocking capabilities. > Element has no indexing capabilities. > Element has no URI handling capabilities. > > Pads: > SRC: 'src' > Implementation: > Has custom eventfunc(): gst_pad_event_default > Has custom queryfunc(): gst_pad_query_default > Provides query types: > Has custom intconnfunc(): gst_pad_get_internal_links_default > Pad Template: 'src' > SINK: 'sink' > Implementation: > Has chainfunc(): gst_msgsmdec_chain > Has custom eventfunc(): gst_pad_event_default > Has custom queryfunc(): gst_pad_query_default > Provides query types: > Has custom intconnfunc(): gst_pad_get_internal_links_default > Pad Template: 'sink' > > Element Properties: > name : The name of the object > flags: readable, writable > String. Default: null Current: "msgsmdec0" > > > Michael Smith wrote: >> On Fri, Jan 16, 2009 at 2:08 PM, Tom Vacek wrote: >> >>> A voicemail system (Asterisk) forwards messages to me in "msgsm" >>> format. I can play them with Blastwave mplayer (see below). In a >>> nutshell, I compiled a gstreamer msgsm plugin, and gstreamer appears to >>> recognize it, but my messages still won't play. >>> >>> The following three command outputs tell the story: >>> >> >> Not really. "gst-inspect msgsmdec" would be the crucial missing info... >> >> Mike >> >> >>> ====================================== >>> xx at v1:~$ gst-inspect msgsm >>> Plugin Details: >>> Name: msgsm >>> Description: msGSM encoder/decoder >>> Filename: /usr/lib/gstreamer-0.10/libgstmsgsm.so >>> Version: 0.12.10 >>> License: LGPL >>> Source module: gst-plugins-farsight >>> Binary package: Farsight >>> Origin URL: http://farsight.sf.net >>> >>> msgsmenc: MS GSM audio encoder >>> msgsmdec: MS GSM audio decoder >>> >>> 2 features: >>> +-- 2 elements >>> >>> ============================ >>> xx at v1:~$ totem msg0000.WAV >>> ** Message: don't know how to handle audio/ms-gsm, rate=(int)8000, >>> channels=(int)1, codec_data=(buffer)4001 >>> ** Message: Error: You do not have a decoder installed to handle this >>> file. You might need to install the necessary plugins. >>> gstplaybasebin.c(2307): prepare_output (): /GstPlayBin:play >>> >>> ** Message: Missing plugin: gstreamer|0.10|totem|MS GSM >>> decoder|decoder-audio/ms-gsm (MS GSM decoder) >>> ** Message: Automatic missing codec installation not supported (helper >>> script missing) >>> >>> ======================================= >>> xx at v1:~$ mplayer msg0000.WAV >>> MPlayer 1.0rc2-CSW-Studio11 (C) 2000-2007 MPlayer Team >>> CPU: Intel x86 >>> CPUflags: MMX: 0 MMX2: 0 3DNow: 0 3DNow2: 0 SSE: 0 SSE2: 0 >>> Compiled for x86 CPU with extensions: >>> >>> Playing msg0000.WAV. >>> Audio file file format detected. >>> ========================================================================== >>> Opening audio decoder: [msgsm] native GSM/MSGSM audio decoder >>> AUDIO: 8000 Hz, 1 ch, s16le, 13.0 kbit/10.16% (ratio: 1625->16000) >>> Selected audio codec: [msgsm] afm: msgsm (MS GSM) >>> ========================================================================== >>> >>> >>> >>> >>> >>> >>> ------------------------------------------------------------------------------ >>> This SF.net email is sponsored by: >>> SourcForge Community >>> SourceForge wants to tell your story. >>> http://p.sf.net/sfu/sf-spreadtheword >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> >> ------------------------------------------------------------------------------ >> This SF.net email is sponsored by: >> SourcForge Community >> SourceForge wants to tell your story. >> http://p.sf.net/sfu/sf-spreadtheword >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From vacek008 at umn.edu Sat Jan 17 01:05:27 2009 From: vacek008 at umn.edu (Tom Vacek) Date: Fri, 16 Jan 2009 18:05:27 -0600 Subject: [gst-devel] msgsm plugin not recognized In-Reply-To: <3c1737210901161557i2e166b10yc99018b37bd7f93d@mail.gmail.com> References: <49710548.6020101@umn.edu> <3c1737210901161500o13bbf362ide420f2b8fe8b2d0@mail.gmail.com> <49711257.3000006@umn.edu> <3c1737210901161557i2e166b10yc99018b37bd7f93d@mail.gmail.com> Message-ID: <497120C7.5050307@umn.edu> I was wondering if that was the problem. Do you think it would worthwhile to file a bug against Asterisk for labeling the file incorrectly? What should the name of this format be? Thanks for the pointer. Tom Michael Smith wrote: > Tom, > > So, from your first email we have this:> Capabilities: > >> audio/x-msgsm >> rate: 8000 >> channels: 1 >> > > ** Message: don't know how to handle audio/ms-gsm, rate=(int)8000, > channels=(int)1, codec_data=(buffer)4001 > > >From the gst-inspect output, we have this: > Capabilities: > audio/x-msgsm > rate: 8000 > channels: 1 > > This means that the msgsmdec decoder that you have accepts a type > called "audio/x-msgsm". However, the demuxer outputs a type called > "audio/ms-gsm". These aren't the same, so the decoder is considered > incompatible, and therefore not used. > > You'll need to change the decoder to use "audio/ms-gsm" for this to work. > > Mike > > > > On Fri, Jan 16, 2009 at 3:03 PM, Tom Vacek wrote: > >> Thanks for looking at this. Here it is. >> >> twvacek at v1:~/Desktop$ gst-inspect msgsmdec >> Factory Details: >> Long name: MS GSM audio decoder >> Class: Codec/Decoder/Audio >> Description: Decodes MS GSM encoded audio >> Author(s): Philippe Khalaf >> Rank: none (0) >> >> Plugin Details: >> Name: msgsm >> Description: msGSM encoder/decoder >> Filename: /usr/lib/gstreamer-0.10/libgstmsgsm.so >> Version: 0.12.10 >> License: LGPL >> Source module: gst-plugins-farsight >> Binary package: Farsight >> Origin URL: http://farsight.sf.net >> >> GObject >> +----GstObject >> +----GstElement >> +----GstMSGSMDec >> >> Pad Templates: >> SINK template: 'sink' >> Availability: Always >> Capabilities: >> audio/x-msgsm >> rate: 8000 >> channels: 1 >> >> SRC template: 'src' >> Availability: Always >> Capabilities: >> audio/x-raw-int >> endianness: 1234 >> signed: true >> width: 16 >> depth: 16 >> rate: 8000 >> channels: 1 >> >> >> Element Flags: >> no flags set >> >> Element Implementation: >> Has change_state() function: gst_element_change_state_func >> Has custom save_thyself() function: gst_element_save_thyself >> Has custom restore_thyself() function: gst_element_restore_thyself >> >> Element has no clocking capabilities. >> Element has no indexing capabilities. >> Element has no URI handling capabilities. >> >> Pads: >> SRC: 'src' >> Implementation: >> Has custom eventfunc(): gst_pad_event_default >> Has custom queryfunc(): gst_pad_query_default >> Provides query types: >> Has custom intconnfunc(): gst_pad_get_internal_links_default >> Pad Template: 'src' >> SINK: 'sink' >> Implementation: >> Has chainfunc(): gst_msgsmdec_chain >> Has custom eventfunc(): gst_pad_event_default >> Has custom queryfunc(): gst_pad_query_default >> Provides query types: >> Has custom intconnfunc(): gst_pad_get_internal_links_default >> Pad Template: 'sink' >> >> Element Properties: >> name : The name of the object >> flags: readable, writable >> String. Default: null Current: "msgsmdec0" >> >> >> Michael Smith wrote: >> >>> On Fri, Jan 16, 2009 at 2:08 PM, Tom Vacek wrote: >>> >>> >>>> A voicemail system (Asterisk) forwards messages to me in "msgsm" >>>> format. I can play them with Blastwave mplayer (see below). In a >>>> nutshell, I compiled a gstreamer msgsm plugin, and gstreamer appears to >>>> recognize it, but my messages still won't play. >>>> >>>> The following three command outputs tell the story: >>>> >>>> >>> Not really. "gst-inspect msgsmdec" would be the crucial missing info... >>> >>> Mike >>> >>> >>> >>>> ====================================== >>>> xx at v1:~$ gst-inspect msgsm >>>> Plugin Details: >>>> Name: msgsm >>>> Description: msGSM encoder/decoder >>>> Filename: /usr/lib/gstreamer-0.10/libgstmsgsm.so >>>> Version: 0.12.10 >>>> License: LGPL >>>> Source module: gst-plugins-farsight >>>> Binary package: Farsight >>>> Origin URL: http://farsight.sf.net >>>> >>>> msgsmenc: MS GSM audio encoder >>>> msgsmdec: MS GSM audio decoder >>>> >>>> 2 features: >>>> +-- 2 elements >>>> >>>> ============================ >>>> xx at v1:~$ totem msg0000.WAV >>>> ** Message: don't know how to handle audio/ms-gsm, rate=(int)8000, >>>> channels=(int)1, codec_data=(buffer)4001 >>>> ** Message: Error: You do not have a decoder installed to handle this >>>> file. You might need to install the necessary plugins. >>>> gstplaybasebin.c(2307): prepare_output (): /GstPlayBin:play >>>> >>>> ** Message: Missing plugin: gstreamer|0.10|totem|MS GSM >>>> decoder|decoder-audio/ms-gsm (MS GSM decoder) >>>> ** Message: Automatic missing codec installation not supported (helper >>>> script missing) >>>> >>>> ======================================= >>>> xx at v1:~$ mplayer msg0000.WAV >>>> MPlayer 1.0rc2-CSW-Studio11 (C) 2000-2007 MPlayer Team >>>> CPU: Intel x86 >>>> CPUflags: MMX: 0 MMX2: 0 3DNow: 0 3DNow2: 0 SSE: 0 SSE2: 0 >>>> Compiled for x86 CPU with extensions: >>>> >>>> Playing msg0000.WAV. >>>> Audio file file format detected. >>>> ========================================================================== >>>> Opening audio decoder: [msgsm] native GSM/MSGSM audio decoder >>>> AUDIO: 8000 Hz, 1 ch, s16le, 13.0 kbit/10.16% (ratio: 1625->16000) >>>> Selected audio codec: [msgsm] afm: msgsm (MS GSM) >>>> ========================================================================== >>>> >>>> >>>> >>>> >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> This SF.net email is sponsored by: >>>> SourcForge Community >>>> SourceForge wants to tell your story. >>>> http://p.sf.net/sfu/sf-spreadtheword >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> gstreamer-devel at lists.sourceforge.net >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>>> >>>> >>> ------------------------------------------------------------------------------ >>> This SF.net email is sponsored by: >>> SourcForge Community >>> SourceForge wants to tell your story. >>> http://p.sf.net/sfu/sf-spreadtheword >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> ------------------------------------------------------------------------------ >> This SF.net email is sponsored by: >> SourcForge Community >> SourceForge wants to tell your story. >> http://p.sf.net/sfu/sf-spreadtheword >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From msmith at xiph.org Sat Jan 17 01:08:39 2009 From: msmith at xiph.org (Michael Smith) Date: Fri, 16 Jan 2009 16:08:39 -0800 Subject: [gst-devel] msgsm plugin not recognized In-Reply-To: <497120C7.5050307@umn.edu> References: <49710548.6020101@umn.edu> <3c1737210901161500o13bbf362ide420f2b8fe8b2d0@mail.gmail.com> <49711257.3000006@umn.edu> <3c1737210901161557i2e166b10yc99018b37bd7f93d@mail.gmail.com> <497120C7.5050307@umn.edu> Message-ID: <3c1737210901161608w1ba9f54em2f8713e0d67d2cc2@mail.gmail.com> On Fri, Jan 16, 2009 at 4:05 PM, Tom Vacek wrote: > I was wondering if that was the problem. Do you think it would > worthwhile to file a bug against Asterisk for labeling the file > incorrectly? What should the name of this format be? Asterisk appears to be doing the right thing - the gstreamer caps types are internal to gstreamer. So it's the gstreamer "msgsm" plugin that's doing the wrong thing, nothing else. Mike From felipe.contreras at gmail.com Sat Jan 17 11:38:55 2009 From: felipe.contreras at gmail.com (Felipe Contreras) Date: Sat, 17 Jan 2009 12:38:55 +0200 Subject: [gst-devel] libmpg123 decoder plugin In-Reply-To: <20090115164447.GF14461@burning-bastards.com> References: <20090114154116.GZ14461@burning-bastards.com> <94a0d4530901142147w600c113fu1b5d2cc5844b4929@mail.gmail.com> <20090115123521.GD14461@burning-bastards.com> <94a0d4530901150506i3cd7ffb7ydb8d829e20966d66@mail.gmail.com> <20090115164447.GF14461@burning-bastards.com> Message-ID: <94a0d4530901170238y2e723391t224e2b04e6667ab1@mail.gmail.com> On Thu, Jan 15, 2009 at 6:44 PM, Roland Poppenreiter wrote: > * On Thu, 15 Jan 2009 at 15:06 +0200, Felipe Contreras wrote: >> But that's the point, if you provide the mp123 library with OpenMAX IL >> interface, then you don't need to write your decoder element. > > Sorry I dont get it, why should I use the mpg123 library with > OpenMAX interface if I already have a working implementation of the > mpg123 decoder plugin. Now that you have your mpg123 plugin working it probably doesn't make much sense. Ideally other multimedia frameworks would have OpenMAX IL support, and so if both mad and libmpg123 provide an OpenMAX IL API they would work transparently on the same multimedia frameworks [1]. That's of course theoretical because so far I only know of GStreamer, Helix, and Android's player that support OpenMAX IL. [1] http://www.freedesktop.org/wiki/GstOpenMAX?action=AttachFile&do=get&target=gst-openmax.png -- Felipe Contreras From thijsvermeir at gmail.com Sat Jan 17 16:35:31 2009 From: thijsvermeir at gmail.com (Thijs Vermeir) Date: Sat, 17 Jan 2009 16:35:31 +0100 Subject: [gst-devel] msgsm plugin not recognized In-Reply-To: <49711257.3000006@umn.edu> References: <49710548.6020101@umn.edu> <3c1737210901161500o13bbf362ide420f2b8fe8b2d0@mail.gmail.com> <49711257.3000006@umn.edu> Message-ID: <20090117153531.GA24021@TVFluendoHome> Hi, On Fri, Jan 16, 2009 at 05:03:51PM -0600, Tom Vacek wrote: > Thanks for looking at this. Here it is. > > twvacek at v1:~/Desktop$ gst-inspect msgsmdec > Factory Details: > Long name: MS GSM audio decoder > Class: Codec/Decoder/Audio > Description: Decodes MS GSM encoded audio > Author(s): Philippe Khalaf > Rank: none (0) Also, decoders with a rank 0 will not be auto-plugged. This needs to be higher than 0 before Totem will use this decoder. Gr, Thijs > > Plugin Details: > Name: msgsm > Description: msGSM encoder/decoder > Filename: /usr/lib/gstreamer-0.10/libgstmsgsm.so > Version: 0.12.10 > License: LGPL > Source module: gst-plugins-farsight > Binary package: Farsight > Origin URL: http://farsight.sf.net > > GObject > +----GstObject > +----GstElement > +----GstMSGSMDec > > Pad Templates: > SINK template: 'sink' > Availability: Always > Capabilities: > audio/x-msgsm > rate: 8000 > channels: 1 > > SRC template: 'src' > Availability: Always > Capabilities: > audio/x-raw-int > endianness: 1234 > signed: true > width: 16 > depth: 16 > rate: 8000 > channels: 1 > > > Element Flags: > no flags set > > Element Implementation: > Has change_state() function: gst_element_change_state_func > Has custom save_thyself() function: gst_element_save_thyself > Has custom restore_thyself() function: gst_element_restore_thyself > > Element has no clocking capabilities. > Element has no indexing capabilities. > Element has no URI handling capabilities. > > Pads: > SRC: 'src' > Implementation: > Has custom eventfunc(): gst_pad_event_default > Has custom queryfunc(): gst_pad_query_default > Provides query types: > Has custom intconnfunc(): gst_pad_get_internal_links_default > Pad Template: 'src' > SINK: 'sink' > Implementation: > Has chainfunc(): gst_msgsmdec_chain > Has custom eventfunc(): gst_pad_event_default > Has custom queryfunc(): gst_pad_query_default > Provides query types: > Has custom intconnfunc(): gst_pad_get_internal_links_default > Pad Template: 'sink' > > Element Properties: > name : The name of the object > flags: readable, writable > String. Default: null Current: "msgsmdec0" > > > Michael Smith wrote: > > On Fri, Jan 16, 2009 at 2:08 PM, Tom Vacek wrote: > > > >> A voicemail system (Asterisk) forwards messages to me in "msgsm" > >> format. I can play them with Blastwave mplayer (see below). In a > >> nutshell, I compiled a gstreamer msgsm plugin, and gstreamer appears to > >> recognize it, but my messages still won't play. > >> > >> The following three command outputs tell the story: > >> > > > > Not really. "gst-inspect msgsmdec" would be the crucial missing info... > > > > Mike > > > > > >> ====================================== > >> xx at v1:~$ gst-inspect msgsm > >> Plugin Details: > >> Name: msgsm > >> Description: msGSM encoder/decoder > >> Filename: /usr/lib/gstreamer-0.10/libgstmsgsm.so > >> Version: 0.12.10 > >> License: LGPL > >> Source module: gst-plugins-farsight > >> Binary package: Farsight > >> Origin URL: http://farsight.sf.net > >> > >> msgsmenc: MS GSM audio encoder > >> msgsmdec: MS GSM audio decoder > >> > >> 2 features: > >> +-- 2 elements > >> > >> ============================ > >> xx at v1:~$ totem msg0000.WAV > >> ** Message: don't know how to handle audio/ms-gsm, rate=(int)8000, > >> channels=(int)1, codec_data=(buffer)4001 > >> ** Message: Error: You do not have a decoder installed to handle this > >> file. You might need to install the necessary plugins. > >> gstplaybasebin.c(2307): prepare_output (): /GstPlayBin:play > >> > >> ** Message: Missing plugin: gstreamer|0.10|totem|MS GSM > >> decoder|decoder-audio/ms-gsm (MS GSM decoder) > >> ** Message: Automatic missing codec installation not supported (helper > >> script missing) > >> > >> ======================================= > >> xx at v1:~$ mplayer msg0000.WAV > >> MPlayer 1.0rc2-CSW-Studio11 (C) 2000-2007 MPlayer Team > >> CPU: Intel x86 > >> CPUflags: MMX: 0 MMX2: 0 3DNow: 0 3DNow2: 0 SSE: 0 SSE2: 0 > >> Compiled for x86 CPU with extensions: > >> > >> Playing msg0000.WAV. > >> Audio file file format detected. > >> ========================================================================== > >> Opening audio decoder: [msgsm] native GSM/MSGSM audio decoder > >> AUDIO: 8000 Hz, 1 ch, s16le, 13.0 kbit/10.16% (ratio: 1625->16000) > >> Selected audio codec: [msgsm] afm: msgsm (MS GSM) > >> ========================================================================== > >> > >> > >> > >> > >> > >> > >> ------------------------------------------------------------------------------ > >> This SF.net email is sponsored by: > >> SourcForge Community > >> SourceForge wants to tell your story. > >> http://p.sf.net/sfu/sf-spreadtheword > >> _______________________________________________ > >> gstreamer-devel mailing list > >> gstreamer-devel at lists.sourceforge.net > >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > >> > >> > > > > ------------------------------------------------------------------------------ > > This SF.net email is sponsored by: > > SourcForge Community > > SourceForge wants to tell your story. > > http://p.sf.net/sfu/sf-spreadtheword > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From vacek008 at umn.edu Sat Jan 17 16:45:26 2009 From: vacek008 at umn.edu (Tom Vacek) Date: Sat, 17 Jan 2009 09:45:26 -0600 Subject: [gst-devel] msgsm plugin not recognized In-Reply-To: <20090117153531.GA24021@TVFluendoHome> References: <49710548.6020101@umn.edu> <3c1737210901161500o13bbf362ide420f2b8fe8b2d0@mail.gmail.com> <49711257.3000006@umn.edu> <20090117153531.GA24021@TVFluendoHome> Message-ID: <4971FD16.8060802@umn.edu> I figured that out eventually. But in the meantime, I discovered that the gsm plugin in gst-plugins-bad handles the msgsm as well as gsm, so I built that instead without a hitch. Just out of curiosity, how does one go about changing the rank? Tom Thijs Vermeir wrote: > Hi, > > On Fri, Jan 16, 2009 at 05:03:51PM -0600, Tom Vacek wrote: > >> Thanks for looking at this. Here it is. >> >> twvacek at v1:~/Desktop$ gst-inspect msgsmdec >> Factory Details: >> Long name: MS GSM audio decoder >> Class: Codec/Decoder/Audio >> Description: Decodes MS GSM encoded audio >> Author(s): Philippe Khalaf >> Rank: none (0) >> > > Also, decoders with a rank 0 will not be auto-plugged. This needs to be > higher than 0 before Totem will use this decoder. > > Gr, > Thijs > > >> Plugin Details: >> Name: msgsm >> Description: msGSM encoder/decoder >> Filename: /usr/lib/gstreamer-0.10/libgstmsgsm.so >> Version: 0.12.10 >> License: LGPL >> Source module: gst-plugins-farsight >> Binary package: Farsight >> Origin URL: http://farsight.sf.net >> >> GObject >> +----GstObject >> +----GstElement >> +----GstMSGSMDec >> >> Pad Templates: >> SINK template: 'sink' >> Availability: Always >> Capabilities: >> audio/x-msgsm >> rate: 8000 >> channels: 1 >> >> SRC template: 'src' >> Availability: Always >> Capabilities: >> audio/x-raw-int >> endianness: 1234 >> signed: true >> width: 16 >> depth: 16 >> rate: 8000 >> channels: 1 >> >> >> Element Flags: >> no flags set >> >> Element Implementation: >> Has change_state() function: gst_element_change_state_func >> Has custom save_thyself() function: gst_element_save_thyself >> Has custom restore_thyself() function: gst_element_restore_thyself >> >> Element has no clocking capabilities. >> Element has no indexing capabilities. >> Element has no URI handling capabilities. >> >> Pads: >> SRC: 'src' >> Implementation: >> Has custom eventfunc(): gst_pad_event_default >> Has custom queryfunc(): gst_pad_query_default >> Provides query types: >> Has custom intconnfunc(): gst_pad_get_internal_links_default >> Pad Template: 'src' >> SINK: 'sink' >> Implementation: >> Has chainfunc(): gst_msgsmdec_chain >> Has custom eventfunc(): gst_pad_event_default >> Has custom queryfunc(): gst_pad_query_default >> Provides query types: >> Has custom intconnfunc(): gst_pad_get_internal_links_default >> Pad Template: 'sink' >> >> Element Properties: >> name : The name of the object >> flags: readable, writable >> String. Default: null Current: "msgsmdec0" >> >> >> Michael Smith wrote: >> >>> On Fri, Jan 16, 2009 at 2:08 PM, Tom Vacek wrote: >>> >>> >>>> A voicemail system (Asterisk) forwards messages to me in "msgsm" >>>> format. I can play them with Blastwave mplayer (see below). In a >>>> nutshell, I compiled a gstreamer msgsm plugin, and gstreamer appears to >>>> recognize it, but my messages still won't play. >>>> >>>> The following three command outputs tell the story: >>>> >>>> >>> Not really. "gst-inspect msgsmdec" would be the crucial missing info... >>> >>> Mike >>> >>> >>> >>>> ====================================== >>>> xx at v1:~$ gst-inspect msgsm >>>> Plugin Details: >>>> Name: msgsm >>>> Description: msGSM encoder/decoder >>>> Filename: /usr/lib/gstreamer-0.10/libgstmsgsm.so >>>> Version: 0.12.10 >>>> License: LGPL >>>> Source module: gst-plugins-farsight >>>> Binary package: Farsight >>>> Origin URL: http://farsight.sf.net >>>> >>>> msgsmenc: MS GSM audio encoder >>>> msgsmdec: MS GSM audio decoder >>>> >>>> 2 features: >>>> +-- 2 elements >>>> >>>> ============================ >>>> xx at v1:~$ totem msg0000.WAV >>>> ** Message: don't know how to handle audio/ms-gsm, rate=(int)8000, >>>> channels=(int)1, codec_data=(buffer)4001 >>>> ** Message: Error: You do not have a decoder installed to handle this >>>> file. You might need to install the necessary plugins. >>>> gstplaybasebin.c(2307): prepare_output (): /GstPlayBin:play >>>> >>>> ** Message: Missing plugin: gstreamer|0.10|totem|MS GSM >>>> decoder|decoder-audio/ms-gsm (MS GSM decoder) >>>> ** Message: Automatic missing codec installation not supported (helper >>>> script missing) >>>> >>>> ======================================= >>>> xx at v1:~$ mplayer msg0000.WAV >>>> MPlayer 1.0rc2-CSW-Studio11 (C) 2000-2007 MPlayer Team >>>> CPU: Intel x86 >>>> CPUflags: MMX: 0 MMX2: 0 3DNow: 0 3DNow2: 0 SSE: 0 SSE2: 0 >>>> Compiled for x86 CPU with extensions: >>>> >>>> Playing msg0000.WAV. >>>> Audio file file format detected. >>>> ========================================================================== >>>> Opening audio decoder: [msgsm] native GSM/MSGSM audio decoder >>>> AUDIO: 8000 Hz, 1 ch, s16le, 13.0 kbit/10.16% (ratio: 1625->16000) >>>> Selected audio codec: [msgsm] afm: msgsm (MS GSM) >>>> ========================================================================== >>>> >>>> >>>> >>>> >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> This SF.net email is sponsored by: >>>> SourcForge Community >>>> SourceForge wants to tell your story. >>>> http://p.sf.net/sfu/sf-spreadtheword >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> gstreamer-devel at lists.sourceforge.net >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>>> >>>> >>> ------------------------------------------------------------------------------ >>> This SF.net email is sponsored by: >>> SourcForge Community >>> SourceForge wants to tell your story. >>> http://p.sf.net/sfu/sf-spreadtheword >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> ------------------------------------------------------------------------------ >> This SF.net email is sponsored by: >> SourcForge Community >> SourceForge wants to tell your story. >> http://p.sf.net/sfu/sf-spreadtheword >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From wim.taymans at gmail.com Sat Jan 17 16:49:11 2009 From: wim.taymans at gmail.com (Wim Taymans) Date: Sat, 17 Jan 2009 16:49:11 +0100 Subject: [gst-devel] msgsm plugin not recognized In-Reply-To: <49710548.6020101@umn.edu> References: <49710548.6020101@umn.edu> Message-ID: <1232207352.27492.7.camel@metal> On Fri, 2009-01-16 at 16:08 -0600, Tom Vacek wrote: > A voicemail system (Asterisk) forwards messages to me in "msgsm" > format. I can play them with Blastwave mplayer (see below). In a > nutshell, I compiled a gstreamer msgsm plugin, and gstreamer appears to > recognize it, but my messages still won't play. > > The following three command outputs tell the story: > > ====================================== > xx at v1:~$ gst-inspect msgsm > Plugin Details: > Name: msgsm > Description: msGSM encoder/decoder > Filename: /usr/lib/gstreamer-0.10/libgstmsgsm.so > Version: 0.12.10 > License: LGPL > Source module: gst-plugins-farsight > Binary package: Farsight > Origin URL: http://farsight.sf.net > > msgsmenc: MS GSM audio encoder > msgsmdec: MS GSM audio decoder > > 2 features: > +-- 2 elements > > ============================ > xx at v1:~$ totem msg0000.WAV > ** Message: don't know how to handle audio/ms-gsm, rate=(int)8000, > channels=(int)1, codec_data=(buffer)4001 > ** Message: Error: You do not have a decoder installed to handle this > file. You might need to install the necessary plugins. > gstplaybasebin.c(2307): prepare_output (): /GstPlayBin:play > > ** Message: Missing plugin: gstreamer|0.10|totem|MS GSM > decoder|decoder-audio/ms-gsm (MS GSM decoder) > ** Message: Automatic missing codec installation not supported (helper > script missing) You need the gsm decoder from gst-plugins-bad to play this file. Wim > > ======================================= > xx at v1:~$ mplayer msg0000.WAV > MPlayer 1.0rc2-CSW-Studio11 (C) 2000-2007 MPlayer Team > CPU: Intel x86 > CPUflags: MMX: 0 MMX2: 0 3DNow: 0 3DNow2: 0 SSE: 0 SSE2: 0 > Compiled for x86 CPU with extensions: > > Playing msg0000.WAV. > Audio file file format detected. > ========================================================================== > Opening audio decoder: [msgsm] native GSM/MSGSM audio decoder > AUDIO: 8000 Hz, 1 ch, s16le, 13.0 kbit/10.16% (ratio: 1625->16000) > Selected audio codec: [msgsm] afm: msgsm (MS GSM) > ========================================================================== > > > > > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From julien.isorce at gmail.com Sat Jan 17 16:58:01 2009 From: julien.isorce at gmail.com (Julien Isorce) Date: Sat, 17 Jan 2009 16:58:01 +0100 Subject: [gst-devel] New GStreamer binaries for Windows based on OAbuild. In-Reply-To: <772db3280901070342sfd0de76j9ca1169e29116972@mail.gmail.com> References: <772db3280901070342sfd0de76j9ca1169e29116972@mail.gmail.com> Message-ID: <180a127d0901170758w7a507777ha592735f2fb7c1da@mail.gmail.com> Hi, I was trying to use your packages ... but You are using glib 2.16.1 according to the "glibconfig.h" which is deployed with your installers (http://forja.rediris.es/frs/?group_id=96) In this doc page http://library.gnome.org/devel/glib/stable/glib-Threads.html#g-once-init-enterwe can read that there is a function called g_once_init_enter since glib 2.14. I go into "gthread.h" (which is deployed with your installers) I cannot see any g_once_init_enter definition. It causes a failure when using gstutils.h: "#if GLIB_CHECK_VERSION (2, 14, 0) #define __gst_once_init_enter(val) (g_once_init_enter (val)) #define __gst_once_init_leave(val,newval) (g_once_init_leave (val, newval)) #else #define __gst_once_init_enter(val) (G_UNLIKELY (*(val) == 0)) #define __gst_once_init_leave(val,newval) (*(val) = newval) #endif " Any explanations ? Sincerely Julien Le 7 janvier 2009 12:42, Andoni Morales a ?crit : > During the last couple of months we have been working on porting GStreamer > to Windows using the OAbuild project. We introduced some modifications to > OAbuild in order to improve compatibility and provide video decoding and > encoding in a larger number of codecs. > > Changelog: > ?Compile ffmpeg with mingw/msys with full codec support. > ?Remove the ffmpeg project from OAbuild and link the ffmpeg plugin to the > mingw/msys ffmpeg libraries. > ?Remove the GLib project from OAbuild and link GStreamer to the GLib > version provided > with the GTK+ win32 installer (in this way GStreamer can be used by GTK > applications). > ?Update liboil to a newer version to compile the volume plugin. > ?Change the ffmpeg plugin code to compile it using VS. > ?Change the qtdemux plugin code to compile it using VS. > ?Compile libmad for Windows and add the libraries to the Gstreamer's > dependencies to provide mp3 decoding support. > ?Add the mad plugin to OAbuild. > ?Add the asfdemux plugin to OAbuild. > ?Remove the Direct Show wrapper and use the ffmepg plugin for all the > decoding stuff. > ?Add the xvid plugin to OAbuild. > ?Change the xvid plugin code to complile it usin VS. > ?Add the mpeg2decoder plugin to OAbuild. > ?Add the mpegstream plugin to OAbuild. > > To do: > ?Make the x264 plugin work > ?Improve mpeg2 decoding/demuxing support > ?Add dvd navigation support > > > You can find the installers at: > http://forja.rediris.es/frs/?group_id=96 > > > Andoni Morales Alastruey > Andr?s Colubri > > > ------------------------------------------------------------------------------ > Check out the new SourceForge.net Marketplace. > It is the best place to buy or sell services for > just about anything Open Source. > http://p.sf.net/sfu/Xq1LFB > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jpuydt at free.fr Sat Jan 17 21:00:42 2009 From: jpuydt at free.fr (Julien Puydt) Date: Sat, 17 Jan 2009 21:00:42 +0100 Subject: [gst-devel] I can't get an appsrc-using pipeline to PLAYING state In-Reply-To: <496ED0FE.4070003@free.fr> References: <496C4AD0.10504@free.fr> <496E48C9.7070106@free.fr> <496E51D6.6020109@free.fr> <496ED0FE.4070003@free.fr> Message-ID: <497238EA.2000407@free.fr> Julien Puydt a ?crit : > Julien Puydt a ?crit : >> Sigh... the pipeline missed a " ! " -- now it goes into PLAYING state, I >> see buffers getting pushed to it... but no sound. > > I replaced the sdlaudiosink with a filesink : empty file -- the buffers > don't get through! The pipeline is now : appsrc is-live=true name=ekiga_src caps=audio/x-raw-int,rate=44100,channels=2,width=16 ! volume name=ekiga_volume ! pulsesink device=combined But it fails with : ** ** ERROR:(gstaudiofilter.c:167):gst_audio_filter_set_caps: assertion failed: (gst_caps_is_equal (incaps, outcaps)) what happens? Snark on #gstreamer From thaytan at noraisin.net Sun Jan 18 00:24:59 2009 From: thaytan at noraisin.net (Jan Schmidt) Date: Sat, 17 Jan 2009 23:24:59 +0000 Subject: [gst-devel] 4th pre-releases of Core/Base available. Bad/Python unchanged. In-Reply-To: <1231886995.1445.6.camel@fancy-ubuntu> References: <1231279326.6539.5.camel@fancy-ubuntu> <1231286471.6539.17.camel@fancy-ubuntu> <1231544503.21410.4.camel@fancy-ubuntu> <1231886995.1445.6.camel@fancy-ubuntu> Message-ID: <1232234699.1311.3.camel@fancy-ubuntu> Hi, Just uploaded 2 new pre-releases of core and base that fix some small bugs. The final releases are still planned for Monday. Current pre-releases for each of the 4 modules are at: http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.21.4.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.21.4.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-bad/pre/gst-plugins-bad-0.10.9.3.tar.bz2 and http://gstreamer.freedesktop.org/src/gst-python/pre/gst-python-0.10.13.3.tar.bz2 Cheers, Jan. On Tue, 2009-01-13 at 22:49 +0000, Jan Schmidt wrote: > Hiyas, > > New pre-releeases of Core and Base are available at > > http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.21.4.tar.bz2 > http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.21.4.tar.bz2 > > These fix a few cosmetic issues. gst-plugins-bad and gst-python are still at > > http://gstreamer.freedesktop.org/src/gst-plugins-bad/pre/gst-plugins-bad-0.10.9.3.tar.bz2 > and > http://gstreamer.freedesktop.org/src/gst-python/pre/gst-python-0.10.13.3.tar.bz2 > > The releases are due out Monday 19th January, unless new blocker bugs are found and filed. > > Cheers, > Jan. > > On Fri, 2009-01-09 at 23:41 +0000, Jan Schmidt wrote: > > Hi all, > > > > A new set of pre-releases of all 4 modules are now available at: > > > > http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.21.3.tar.bz2 > > http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.21.3.tar.bz2 > > http://gstreamer.freedesktop.org/src/gst-plugins-bad/pre/gst-plugins-bad-0.10.9.3.tar.bz2 > > and > > http://gstreamer.freedesktop.org/src/gst-python/pre/gst-python-0.10.13.3.tar.bz2 > > > > These fix a variety of small bugs and omissions. Please give them some > > testing love. File bugs at http://bugzilla.gnome.org > > > > If necessary, there'll be new pre-releases on Tuesday (13 Jan). The > > final release tarballs are expected on Monday week, the 19th January. > > > > After that, we'll be staying frozen for a bit while we switch all the > > repositories over to git. > > > > Cheers, > > Jan. > > > > On Wed, 2009-01-07 at 00:01 +0000, Jan Schmidt wrote: > > > Hi all, > > > > > > The first pre-releases of the 4 modules are now available at: > > > http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.21.2.tar.bz2 > > > http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.21.2.tar.bz2 > > > http://gstreamer.freedesktop.org/src/gst-plugins-bad/pre/gst-plugins-bad-0.10.9.2.tar.bz2 > > > and > > > http://gstreamer.freedesktop.org/src/gst-python/pre/gst-python-0.10.13.2.tar.bz2 > > > > > > Please test, and file any bugs at http://bugzilla.gnome.org/ ! > > > > > > Expect new pre-releases on Friday if needed. Check the release plan to > > > find out about the release schedule: > > > http://www.gstreamer.net/wiki/ReleasePlanning2009-1 > > > > > > Cheers, > > > Jan. > > > > > > > > > On Tue, 2009-01-06 at 22:02 +0000, Jan Schmidt wrote: > > > > Hi all, > > > > > > > > All 4 of the Core, Base Plugins, Bad Plugins and Python bindings modules > > > > are now frozen for releases. The Bad Plugins aren't normally in this > > > > cycle. They're there this month to handle appsrc and appsink moving from > > > > -bad to -base at last, and audioresample trading places with the much > > > > better speexresample. Speexresample is now in -base, but registers with > > > > the name 'audioresample'. The old audioresample is now in -bad as > > > > 'legacyresample' > > > > > > > > See the release plan for the first few months of this year > > > > (http://www.gstreamer.net/wiki/ReleasePlanning2009-1) for more detail, > > > > and expect the pre-releases later tonight. > > > > > > > > Cheers, > > > > Jan. -- Jan Schmidt From jpuydt at free.fr Sun Jan 18 10:42:37 2009 From: jpuydt at free.fr (Julien Puydt) Date: Sun, 18 Jan 2009 10:42:37 +0100 Subject: [gst-devel] I can't get an appsrc-using pipeline to PLAYING state In-Reply-To: <497238EA.2000407@free.fr> References: <496C4AD0.10504@free.fr> <496E48C9.7070106@free.fr> <496E51D6.6020109@free.fr> <496ED0FE.4070003@free.fr> <497238EA.2000407@free.fr> Message-ID: <4972F98D.1050607@free.fr> Julien Puydt a ?crit : > Julien Puydt a ?crit : >> Julien Puydt a ?crit : >>> Sigh... the pipeline missed a " ! " -- now it goes into PLAYING state, I >>> see buffers getting pushed to it... but no sound. >> I replaced the sdlaudiosink with a filesink : empty file -- the buffers >> don't get through! > > The pipeline is now : > appsrc is-live=true name=ekiga_src > caps=audio/x-raw-int,rate=44100,channels=2,width=16 > ! volume name=ekiga_volume > ! pulsesink device=combined > > But it fails with : > ** > ** ERROR:(gstaudiofilter.c:167):gst_audio_filter_set_caps: assertion > failed: (gst_caps_is_equal (incaps, outcaps)) > > what happens? Sigh... here is the trace : #0 0xb7fe3424 in __kernel_vsyscall () #1 0xb653b640 in raise () from /lib/i686/cmov/libc.so.6 #2 0xb653d018 in abort () from /lib/i686/cmov/libc.so.6 #3 0xb6800ce2 in IA__g_assertion_message (domain=0xb683d51e "", file=0xb390a268 "gstaudiofilter.c", line=167, func=0xb390a329 "gst_audio_filter_set_caps", message=0x8a57a40 "assertion failed: (gst_caps_is_equal (incaps, outcaps))") at /tmp/buildd/glib2.0-2.16.6/glib/gtestutils.c:1218 #4 0xb680120d in IA__g_assertion_message_expr (domain=0x0, file=0xb390a268 "gstaudiofilter.c", line=167, func=0xb390a329 "gst_audio_filter_set_caps", expr=0xb390a2d0 "gst_caps_is_equal (incaps, outcaps)") at /tmp/buildd/glib2.0-2.16.6/glib/gtestutils.c:1229 #5 0xb38f7472 in gst_audio_filter_set_caps (btrans=0x8d400a8, incaps=0x8a56fa0, outcaps=0x8b653c0) at gstaudiofilter.c:167 #6 0xb622ab52 in gst_base_transform_configure_caps (trans=0x8d400a8, in=0x8a56fa0, out=0x8b653c0) at gstbasetransform.c:689 #7 0xb622c5ca in gst_base_transform_setcaps (pad=0x8b6a570, caps=0x8a56fa0) at gstbasetransform.c:1099 #8 0xb7ed01ba in gst_pad_set_caps (pad=0x8b6a570, caps=0x8a56fa0) at gstpad.c:2426 #9 0xb7ed0dfc in gst_pad_configure_sink (pad=0x8b6a570, caps=0x8a56fa0) at gstpad.c:2480 #10 0xb7ed1063 in gst_pad_chain_unchecked (pad=0x8b6a570, buffer=0x84deac8) at gstpad.c:3875 #11 0xb7ed236a in gst_pad_push (pad=0x8b6a4b0, buffer=0x84deac8) at gstpad.c:4057 #12 0xb6226f91 in gst_base_src_loop (pad=0x8b6a4b0) at gstbasesrc.c:2275 #13 0xb7ef39e3 in gst_task_func (task=0x89ede88, tclass=0x82fba08) at gsttask.c:192 #14 0xb68057e6 in g_thread_pool_thread_proxy (data=0x82fba98) at /tmp/buildd/glib2.0-2.16.6/glib/gthreadpool.c:265 #15 0xb680414f in g_thread_create_proxy (data=0x8b60b08) at /tmp/buildd/glib2.0-2.16.6/glib/gthread.c:635 #16 0xb72154c0 in start_thread () from /lib/i686/cmov/libpthread.so.0 #17 0xb65f06de in clone () from /lib/i686/cmov/libc.so.6 Snark From sviluppo at audiias.com Sun Jan 18 17:15:41 2009 From: sviluppo at audiias.com (=?UTF-8?B?UGllcnBhb2xvIGd1bGzCiGE=?=) Date: Sun, 18 Jan 2009 17:15:41 +0100 Subject: [gst-devel] Streaming FLV files in http Message-ID: <497355AD.3020005@audiias.com> Hi guys , this is my first messagge on the Gstreamer's mailing list. SO i try to explain my need. I want to develop an application that take and. mpg files (encoded with ffmpeg , with mpeg2video for the video and mp2 for the audio). I want one files , and encoding to the flv on the fly and streaming in http this flv I see that with gstreamer it's possible , but I dont' know the command line instructions that I must to use on my Ubuntu 8.0.4 Box. Can you provide to me some links or some instructions? Best Regards. Goodbye Pierpaolo From bisht.sudarshan at gmail.com Mon Jan 19 05:38:37 2009 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Mon, 19 Jan 2009 10:08:37 +0530 Subject: [gst-devel] aacparser - bitrate info Message-ID: <785339900901182038t73b7cb61ybf5d3b063332ce00@mail.gmail.com> Hi all, I am trying to parse an aac file using aacparse plugin , but it seems it does not give correct bitrate information , any idea ? -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From luca.ognibene at gmail.com Mon Jan 19 11:10:45 2009 From: luca.ognibene at gmail.com (Luca Ognibene) Date: Mon, 19 Jan 2009 11:10:45 +0100 Subject: [gst-devel] info about dynamic request pads Message-ID: <2dc5f8a50901190210g7cf16b3fndcae1c2d4cf206f1@mail.gmail.com> Hi, i would like to create a bin that works this way: * a single sink pad * src%d are request pad * inside the bin i have the sink pad linked to a tee * every time i request a pad i want my bin to create a "queue ! identity" bin and attach it to the internal tee * every time i release a pad i want my bin to stop the "queue ! identity" bin and release the internal tee pad Everything work if the bin is in the NULL state but i have segfaults/hangs if the bin is in PLAYING state. I've tried to read part-block and other docs but i'm not able to fix this problem.. I've implemented the bin this way: * in _init the internal tee is created * test app call bin.get_request_pad('src0') * in _request_new_pad: * get element state * request a pad from internal tee * block this pad if state is PLAYING * add the "queue ! identity" bin * create the ghost pad and set target to bin.src * link tee request pad to bin.sink * call sync_state_with_parent on the bin (is this ok? right now bin.src is not linked and the bin is in playing state) * unblock tee request pad * test app block received ghost pad * add queue ! fakesink * link them to ghost pad * call sync_state_with_parent on queue and fakesink * unblock ghost pad Problems: * sometime (really often) program hangs on blocking ghost pad * sometime (not very often) i have segfaults while queue is trying to fixate caps A strange thing is that if i change my bin to add "identity ! identity" instead of "queue ! identity" everything works fine.. but i don't know why! And i need the queue there :) This is a svg graph of my pipeline after having received the ghost pad: http://people.freedesktop.org/~skaboy/test76-3.dot.svg I'm using gstreamer 0.10.21 (core). ciao Luca From thx1138 at earthlink.net Tue Jan 6 00:56:52 2009 From: thx1138 at earthlink.net (thx1138) Date: Mon, 05 Jan 2009 15:56:52 -0800 Subject: [gst-devel] VST api and compatibility with Gstreamer Message-ID: Hello, We are presently using VST as a plug-in for audio applications and the question came up today is it possible for VST plug-ins to work with G-Streamer? Our VST's are migrating from PC/Mac Intel host and desire to use VST approach with Gstreamer for ARM Cortex A8 processor. Regards, Terry From cruxic at gmail.com Wed Jan 7 06:37:13 2009 From: cruxic at gmail.com (Adam B) Date: Tue, 6 Jan 2009 21:37:13 -0800 Subject: [gst-devel] [gst-daily] How to create raw video In-Reply-To: <20090107013858.GA4332@entropywave.com> References: <20090107013858.GA4332@entropywave.com> Message-ID: Thanks for the reply, David. This gets me close because I can bulk convert all my tga images to png before running gst-launch. In that wiki article entry it has me creating an Ogg Theora video file. Is there some lower-level, lossless, file format I can use instead? On Tue, Jan 6, 2009 at 5:38 PM, David Schleef wrote: > On Thu, Jan 01, 2009 at 06:18:27PM -0800, Adam B wrote: > > Hello all, > > > > I have a bunch of TGA image files that I want to convert into a raw, > > uncompressed video. Can I do this with gst-launch? Seems like it would > be > > a simple thing to do but I'm not having much luck with my google > searches. > > It's much simpler if you have PNG images, in which case you can > start from the pipeline in the wiki page: > > http://gstreamer.freedesktop.org/wiki/MultiFileSrc > > > > dave... > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From airteknology at gmail.com Sun Jan 18 17:08:01 2009 From: airteknology at gmail.com (=?UTF-8?B?UGllcnBhb2xvIEd1bGzCiGE=?=) Date: Sun, 18 Jan 2009 17:08:01 +0100 Subject: [gst-devel] Streaming FLV files in http Message-ID: <497353E1.9060409@gmail.com> Hi guys , this is my first messagge on the Gstreamer's mailing list. SO i try to explain my need. I want to develop an application that take and. mpg files (encoded with ffmpeg , with mpeg2video for the video and mp2 for the audio). I want one files , and encoding to the flv on the fly and streaming in http this flv I see that with gstreamer it's possible , but I dont' know the command line instructions that I must to use on my Ubuntu 8.0.4 Box. Can you provide to me some links or some instructions? Best Regards. Goodbye Pierpaolo From rakesh2.sharma at aricent.com Mon Jan 19 12:35:21 2009 From: rakesh2.sharma at aricent.com (rakesh sharma) Date: Mon, 19 Jan 2009 03:35:21 -0800 (PST) Subject: [gst-devel] How to register Gstreamer Plugin with Playbin so that playbin detect my plugins Message-ID: <21541362.post@talk.nabble.com> Hi Friends I have written one plugin now i want that the plugin should be called with playbin.... can anybody please help me its really urgent...!!!! -- View this message in context: http://www.nabble.com/How-to-register-Gstreamer-Plugin-with-Playbin-so-that-playbin-detect-my-plugins-tp21541362p21541362.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From rahul.verma at sasken.com Mon Jan 19 14:59:28 2009 From: rahul.verma at sasken.com (Rahul Verma) Date: Mon, 19 Jan 2009 19:29:28 +0530 Subject: [gst-devel] 3gp demux plugin working on PC but not on target Message-ID: <89635EF2797FED4B9EF9769F80CEABC81AF3C9@EXGZ01.sasken.com> Hi, Sorry for the late response, but was busy with other experiments with gstreamer. Have a few more points to discuss. >> >> Finally I was able to get Audio and Video working on target using my 3gp >> demux. As you pointed out correctly, bug was in my code. I was trying to >> access the file location from filesrc element using the following code, and >> it caused the crash. >> >> fileSrcPeerPad = gst_pad_get_peer(filter->sinkPad); >> if(NULL == fileSrcPeerPad) >> { >> GST_DEBUG("gone.. no src peer pad. "); >> } >> //extract the file src element here and get the filename, fd and uri >> fileSrcPeerElement = gst_pad_get_parent_element(fileSrcPeerPad); >> >> element_class = GST_ELEMENT_CLASS(fileSrcPeerElement); >> >> object_class = G_OBJECT_CLASS(element_class); >> >> //try to extract the location from this GObject. >> g_object_get(object_class, "location", &fileLocation, NULL); >> >> and after that do the file operations. Surprisingly, this piece of code >> worked >> on PC. >> >> Can you help me in finding the error in the code?? >The whole concept is badly broken. Your element should not be poking >at the implementation details of its peer - there's no reason to >expect that the peer even HAS a "location" property. > >So, you should stop doing that entirely. I didn't check the actual >code to see if there was an obvious bug in the actual implementation, >because it really doesn't matter. Thanks Mike. Your point is valid. But for a 3gp reader, which works in a pull mode, the file operations has to be done by the reader, since it knows the parsing logic. If my understanding is correct, then it has to get the file handle. What is the alternative way to get the file handle then? >> >> Also, now I have some AV Sync issues. Till now I have not implemented any AV >> Sync >> logic in my Demux or Sink. I am passing the video buffers decoded to the >> sink, but it >> seems that video frames are dropped by the framework and after that I see >> continuous >> breaks. Does the pipeline implement the frame drop logic? If yes what logic >> does it follow >> for frame drops? can you guide me to the piece of code in the framework for >> a better >> understanding. >Your demuxer should not implement any sync logic - it should timestamp >the buffers it outputs, but nothing more than that. >The sink is responsible for all synchronisation - but in general, you >shouldn't need to implement anything special. The base classes >(GstBaseSink, or subclasses like GstBaseVideoSink) should implement >everything for you - you just need to specify that it's a synchronised >sink. >Frame dropping is also generally done by the sink base classes, if the >buffers arrive too late. Frame dropping SHOULD also be implemented in >decoders (particularly video decoders) in response to QoS messages >from the sinks. The documentation goes into more detail about how this >works (can't remember precisely where, probably the design docs). Thanks again Mike. I went through the implementation of the frame drop logic in gstbasesink.c It is as you said, depending on Timestamp, duration and jitter. I am still facing some AV Sync issues inspite of giving correct timestamp to the video frames. Upon profiling i saw that the Jitter value keeps on increasing, forcing the frames to be dropped. I also profiled the loop function of my demux and was surprised to see that gst_pad_push was taking time in range or 150 ms, specially for audio. Since pad_push is taking time, it increases the jitter and causes the frame drop. my pipeline is filesrc ! 3gpdemux name=demux demux.! queue ! videodecoder ! videosink demux.! queue ! audiodecoder ! osssink Is gst_pad_push a synchronous call and will be blocking? Initially all is well, but as soon as rendering starts, gst_push_pad time starts increasing. rahul SASKEN BUSINESS DISCLAIMER ------------------------- This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, Disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email -------------- next part -------------- An HTML attachment was scrubbed... URL: From arturo at openframeworks.cc Mon Jan 19 14:48:25 2009 From: arturo at openframeworks.cc (arturo castro) Date: Mon, 19 Jan 2009 14:48:25 +0100 Subject: [gst-devel] rate change not very responsive Message-ID: <1232372905.6578.32.camel@art-laptop> Hi I'm one of the developers of the openframeworks project: http://openframeworks.cc , a framework for creative coding. Since now we've been using some object oriented wrapper around FFMPEG as our video backend in the linux version but it's being problematic and super slow so i've decided to give gstreamer a try. With the new version of appsink in base everything has gone more or less smooth except for one little problem. When i try to change a video playing speed from a gui control like a slider or just mapping that property to the mouse position, the video stops playing while speed is changing and only continues playing again when speed stops changing. I'm using a pipeline like: filesrc location=... ! decodebin ! ffmpegcolorspace ! appsink also using signals in appsink to notify new buffers and polling the pipeline for messages with: while(gst_bus_have_pending(bus)){ GstMessage* msg = gst_bus_pop(bus); ... as i cannot get callbacks from the bus since i cannot run in the gstreamer loop because our framework already has it's own. I've tried with diferent seek flags but have more or less the same result. This is the code i'm using to change speed directly mapped to gui control: void ofVideoPlayer::setSpeed(speed) GstFormat format = GST_FORMAT_TIME; GstSeekFlags flags = (GstSeekFlags) (GST_SEEK_FLAG_FLUSH|GST_SEEK_FLAG_ACCURATE); if(!gst_element_seek(GST_ELEMENT(gstPipeline),speed, format, flags, GST_SEEK_TYPE_NONE, 0, GST_SEEK_TYPE_NONE, -1)) { ofLog(OF_WARNING,"GStreamer: unable to seek"); } } also attached is the full code for our videoPlayer class. Is there anything i can do to make speed change more responsive, or is this just a limitation of the current implementation of gstreamer? thanks and keep the good work //------------------------------------------ arturo castro oF core developer barcelona -------------- next part -------------- A non-text attachment was scrubbed... Name: ofVideoPlayerOnlyGst.cpp Type: text/x-c++src Size: 15931 bytes Desc: not available URL: From dstaples at friendmts.co.uk Mon Jan 19 15:47:14 2009 From: dstaples at friendmts.co.uk (Darren Staples) Date: Mon, 19 Jan 2009 14:47:14 +0000 Subject: [gst-devel] Message-ID: I?ve got a Gstreamer pipeline grabbing frames of raw video data several times faster than real-time (which is great !) I?ve done this using something like: while( !gst_app_sink_is_eos( sink )) { buffer = gst_app_sink_pull_buffer( sink ); do_something_with_buffer(...); } This works, but there is no error detection for events other than end of stream. If I were to use a GMainLoop/GMainContext approach instead, is there a Gstreamer event which would tell me every time a frame was ready to be processed ? The GstMessageType documentation seems to hint that GST_MESSAGE_STEP_DONE will do this, but then says it?s not implemented yet... Is there a better way to do this ? Darren *** FRIEND MTS LIMITED EMAIL CONFIDENTIALITY *** This email message and any attachments may contain information which is confidential or privileged and is intended for the sole use of the person to whom it is addressed. If you are not the intended recipient, be aware that any disclosures, copying, distribution or use of the contents is prohibited. If you have received this email message in error, please notify our office by telephone (+44 (0)121 633 2000) or email (admin at friendmts.co.uk) immediately. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: From krishna.kishore at sasken.com Mon Jan 19 15:48:18 2009 From: krishna.kishore at sasken.com (Krishna Kishore) Date: Mon, 19 Jan 2009 20:18:18 +0530 Subject: [gst-devel] Demux References: Message-ID: Hi, I have seen that everyone uses opensource ffmpeg demux or qtdemux. I did not come across anyone writing their own demux using their own parser module. Why does everyone use this opensource demux? Is it because it is already available, well tested? OR did anyone try it and faced problems and switched to opensource demux? Regards, Kishore. SASKEN BUSINESS DISCLAIMER ------------------------- This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, Disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email From arturo at openframeworks.cc Mon Jan 19 16:16:14 2009 From: arturo at openframeworks.cc (arturo castro) Date: Mon, 19 Jan 2009 16:16:14 +0100 Subject: [gst-devel] In-Reply-To: References: Message-ID: <1232378174.6578.39.camel@art-laptop> El lun, 19-01-2009 a las 14:47 +0000, Darren Staples escribi?: > If I were to use a GMainLoop/GMainContext approach instead, is there a > Gstreamer event which would tell me every time a frame was ready to be > processed ? Hi Darren in my code with no GMainLoop i use something like: g_object_set (G_OBJECT (sink), "emit-signals", TRUE, "sync", TRUE, NULL); g_signal_connect (sink, "new-buffer", G_CALLBACK (on_new_buffer_from_source), &gstData); where on_new_buffer_from_source is a callback of the format: static void on_new_buffer_from_source (GstElement * elt, void * data){ guint size; GstBuffer *buffer; //get the buffer from appsink buffer = gst_app_sink_pull_buffer (GST_APP_SINK (elt)); ... that is being notified everytime there's a new buffer to process, and gstData a container to pass the raw pixels to the main app from the callback (in my case needed as the data is inside an object) also if you change "sync", TRUE by "sync", FALSE you'll get the frames at the fastest speed the system can. in your case being raw video i suppose it won' matter anyway ciao //------------------------------------------ arturo castro oF core developer barcelona From msmith at xiph.org Mon Jan 19 19:14:38 2009 From: msmith at xiph.org (Michael Smith) Date: Mon, 19 Jan 2009 10:14:38 -0800 Subject: [gst-devel] VST api and compatibility with Gstreamer In-Reply-To: References: Message-ID: <3c1737210901191014r1c3100e5i4d36318f7f3834b9@mail.gmail.com> On Mon, Jan 5, 2009 at 3:56 PM, thx1138 wrote: > Hello, > > We are presently using VST as a plug-in for audio applications and the > question came up today is it possible for VST plug-ins to work with > G-Streamer? Our VST's are migrating from PC/Mac Intel host and desire to use > VST approach with Gstreamer for ARM Cortex A8 processor. > > Regards, > > Terry It would be possible to build a VST wrapper for gstreamer, and then recompile your VSTs for your ARM platform. As far as I know, nobody has done any of that work though; it might be easier to make your code directly run as gstreamer plugins. Mike From msmith at xiph.org Mon Jan 19 19:16:14 2009 From: msmith at xiph.org (Michael Smith) Date: Mon, 19 Jan 2009 10:16:14 -0800 Subject: [gst-devel] How to register Gstreamer Plugin with Playbin so that playbin detect my plugins In-Reply-To: <21541362.post@talk.nabble.com> References: <21541362.post@talk.nabble.com> Message-ID: <3c1737210901191016s1848ee38q99b52ea8cec1bb2@mail.gmail.com> On Mon, Jan 19, 2009 at 3:35 AM, rakesh sharma wrote: > > Hi Friends I have written one plugin now i want that the plugin should be > called with playbin.... > can anybody please help me its really urgent...!!!! You haven't given much context for us to go on here. Is your plugin a source? A sink? A demuxer or decoder? The key points to check, if you want playbin to autoplug it, are: - correct element details class - non-zero rank (and higher than other alternatives playbin might select) - correct caps templates. We can't be any more specific with the info you've given. Mike From msmith at xiph.org Mon Jan 19 19:24:26 2009 From: msmith at xiph.org (Michael Smith) Date: Mon, 19 Jan 2009 10:24:26 -0800 Subject: [gst-devel] 3gp demux plugin working on PC but not on target In-Reply-To: <89635EF2797FED4B9EF9769F80CEABC81AF3C9@EXGZ01.sasken.com> References: <89635EF2797FED4B9EF9769F80CEABC81AF3C9@EXGZ01.sasken.com> Message-ID: <3c1737210901191024i6232d6d2k89193d22189686bb@mail.gmail.com> On Mon, Jan 19, 2009 at 5:59 AM, Rahul Verma wrote: > >>The whole concept is badly broken. Your element should not be poking >>at the implementation details of its peer - there's no reason to >>expect that the peer even HAS a "location" property. >> >>So, you should stop doing that entirely. I didn't check the actual >>code to see if there was an obvious bug in the actual implementation, >>because it really doesn't matter. > > Thanks Mike. Your point is valid. But for a 3gp reader, which works in a > pull mode, the file operations has to be done by the reader, since it > knows the parsing logic. If my understanding is correct, then it has to get > the file handle. What is the alternative way to get the file handle then? If you're in pull mode, you can (and should) pull from the upstream element - which might, internally, use some sort of 'file handle', but equally, it might not. To do this, you use gst_pad_pull_range() on your element's sinkpad. You MUST NOT directly use any sort of file handle. > > my pipeline is > > filesrc ! 3gpdemux name=demux demux.! queue ! videodecoder ! > videosink demux.! queue ! audiodecoder ! osssink > > Is gst_pad_push a synchronous call and will be blocking? Initially all is > well, but as soon as > rendering starts, gst_push_pad time starts increasing. gst_pad_push() will not generally return immediately. In this pipeline, your demuxer is pushing buffers into a queue element. That should be very fast usually (it just takes the appropriate locks, then adds the buffer to the queue element's internal queue of buffers. However, if the queue is full, it'll block until the queue stops being full. In your case, I don't know why you're seeing increasing time taken for pushing on your pad - perhaps your timestamps or wrong, or your sink or decoder isn't working correctly? Mike From jpuydt at free.fr Mon Jan 19 14:26:55 2009 From: jpuydt at free.fr (Julien Puydt) Date: Mon, 19 Jan 2009 14:26:55 +0100 Subject: [gst-devel] I can't get an appsrc-using pipeline to PLAYING state In-Reply-To: <4972F98D.1050607@free.fr> References: <496C4AD0.10504@free.fr> <496E48C9.7070106@free.fr> <496E51D6.6020109@free.fr> <496ED0FE.4070003@free.fr> <497238EA.2000407@free.fr> <4972F98D.1050607@free.fr> Message-ID: <49747F9F.2060708@free.fr> Julien Puydt a ?crit : >> The pipeline is now : >> appsrc is-live=true name=ekiga_src >> caps=audio/x-raw-int,rate=44100,channels=2,width=16 >> ! volume name=ekiga_volume >> ! pulsesink device=combined >> >> But it fails with : >> ** >> ** ERROR:(gstaudiofilter.c:167):gst_audio_filter_set_caps: assertion >> failed: (gst_caps_is_equal (incaps, outcaps)) >> >> what happens? Search the attached log for the line starting with "Pipeline: " which is ekiga printing what it's trying to do, and look : - I explicitly ask appsrc to have caps caps=audio/x-raw-int,rate=44100,channels=2,width=16 - that doesn't appear in the log (ouch); - what appears in the log is that the pads are connected with caps=(NULL) (ouch ouch). Is it a bug in appsrc or my pipeline? Snark -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: log URL: From xavier.grasa at iss-profesionalia.com Mon Jan 19 19:25:20 2009 From: xavier.grasa at iss-profesionalia.com (xavier grasa) Date: Mon, 19 Jan 2009 19:25:20 +0100 Subject: [gst-devel] New GStreamer binaries for Windows based on OAbuild. In-Reply-To: <180a127d0901170758w7a507777ha592735f2fb7c1da@mail.gmail.com> References: <772db3280901070342sfd0de76j9ca1169e29116972@mail.gmail.com> <180a127d0901170758w7a507777ha592735f2fb7c1da@mail.gmail.com> Message-ID: <4974C590.4050509@iss-profesionalia.com> Hi all, I am trying to build some gstreamer plug-ins under Windows Vista using the installers at (http://forja.rediris.es/frs/?group_id=96), and the same compilation errors appears: error C3861: 'g_once_init_enter': identifier not found regarding g_once_init_enter not being present, just as Julien explained on his mail. Any help would be much appreciated, Best, --Xavier Grasa > Hi, > > I was trying to use your packages ... but > > You are using glib 2.16.1 according to the "glibconfig.h" which is > deployed with your installers (http://forja.rediris.es/frs/?group_id=96) > > In this doc page > http://library.gnome.org/devel/glib/stable/glib-Threads.html#g-once-init-enter > we can read that there is a function called g_once_init_enter since > glib 2.14. > I go into "gthread.h" (which is deployed with your installers) I > cannot see any g_once_init_enter definition. > > It causes a failure when using gstutils.h: > > "#if GLIB_CHECK_VERSION (2, 14, 0) > #define __gst_once_init_enter(val) (g_once_init_enter (val)) > #define __gst_once_init_leave(val,newval) (g_once_init_leave (val, > newval)) > #else > #define __gst_once_init_enter(val) (G_UNLIKELY (*(val) == 0)) > #define __gst_once_init_leave(val,newval) (*(val) = newval) > #endif > " > > Any explanations ? > > Sincerely > > Julien From ylatuya at gmail.com Mon Jan 19 20:25:38 2009 From: ylatuya at gmail.com (Andoni Morales) Date: Mon, 19 Jan 2009 20:25:38 +0100 Subject: [gst-devel] New GStreamer binaries for Windows based on OAbuild. In-Reply-To: <4974C590.4050509@iss-profesionalia.com> References: <772db3280901070342sfd0de76j9ca1169e29116972@mail.gmail.com> <180a127d0901170758w7a507777ha592735f2fb7c1da@mail.gmail.com> <4974C590.4050509@iss-profesionalia.com> Message-ID: <772db3280901191125n136b3834xb4226f7325c66e17@mail.gmail.com> Thanks for the advice... I was using the gtk binaries provided by the mono installer which are a little bit updated. I'll recompile GSStreamer using the binaries provided by the GTK+ team which usesthe 2.18 version of GLib. I think it will be ready for the next week 2009/1/19 xavier grasa > Hi all, > > I am trying to build some gstreamer plug-ins under Windows Vista using > the installers at (http://forja.rediris.es/frs/?group_id=96), and the > same compilation errors appears: > > error C3861: 'g_once_init_enter': identifier not found > > regarding g_once_init_enter not being present, just as Julien explained > on his mail. > > Any help would be much appreciated, > > Best, > > --Xavier Grasa > > > > Hi, > > > > I was trying to use your packages ... but > > > > You are using glib 2.16.1 according to the "glibconfig.h" which is > > deployed with your installers (http://forja.rediris.es/frs/?group_id=96) > > > > In this doc page > > > http://library.gnome.org/devel/glib/stable/glib-Threads.html#g-once-init-enter > > we can read that there is a function called g_once_init_enter since > > glib 2.14. > > I go into "gthread.h" (which is deployed with your installers) I > > cannot see any g_once_init_enter definition. > > > > It causes a failure when using gstutils.h: > > > > "#if GLIB_CHECK_VERSION (2, 14, 0) > > #define __gst_once_init_enter(val) (g_once_init_enter (val)) > > #define __gst_once_init_leave(val,newval) (g_once_init_leave (val, > > newval)) > > #else > > #define __gst_once_init_enter(val) (G_UNLIKELY (*(val) == 0)) > > #define __gst_once_init_leave(val,newval) (*(val) = newval) > > #endif > > " > > > > Any explanations ? > > > > Sincerely > > > > Julien > > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From julien.isorce at gmail.com Mon Jan 19 22:59:23 2009 From: julien.isorce at gmail.com (Julien Isorce) Date: Mon, 19 Jan 2009 22:59:23 +0100 Subject: [gst-devel] New GStreamer binaries for Windows based on OAbuild. In-Reply-To: <4974C590.4050509@iss-profesionalia.com> References: <772db3280901070342sfd0de76j9ca1169e29116972@mail.gmail.com> <180a127d0901170758w7a507777ha592735f2fb7c1da@mail.gmail.com> <4974C590.4050509@iss-profesionalia.com> Message-ID: <180a127d0901191359y528bfce6u36d79f26e5a41320@mail.gmail.com> The solution is to replace all glib .dll, .h, .a, .lib by the last glib packages (2.18.4) from http://www.gtk.org/download-windows.html. So : - install the 2 packages ( http://forja.rediris.es/frs/?group_id=96) - remove include/glib - remove glib .dll(s), .lib and .a (and .def) . I mean keep only gstreamer dev. - unzip glib glib_2.18.4-1_win32.zip and glib-dev_2.18.4-1_win32.zip There is no pb with this because of "acsendent compatibilty" of glib2.x.x. Note I advise you to rebuild everything by your self. Cheers J. 2009/1/19 xavier grasa > Hi all, > > I am trying to build some gstreamer plug-ins under Windows Vista using > the installers at (http://forja.rediris.es/frs/?group_id=96), and the > same compilation errors appears: > > error C3861: 'g_once_init_enter': identifier not found > > regarding g_once_init_enter not being present, just as Julien explained > on his mail. > > Any help would be much appreciated, > > Best, > > --Xavier Grasa > > > > Hi, > > > > I was trying to use your packages ... but > > > > You are using glib 2.16.1 according to the "glibconfig.h" which is > > deployed with your installers (http://forja.rediris.es/frs/?group_id=96) > > > > In this doc page > > > http://library.gnome.org/devel/glib/stable/glib-Threads.html#g-once-init-enter > > we can read that there is a function called g_once_init_enter since > > glib 2.14. > > I go into "gthread.h" (which is deployed with your installers) I > > cannot see any g_once_init_enter definition. > > > > It causes a failure when using gstutils.h: > > > > "#if GLIB_CHECK_VERSION (2, 14, 0) > > #define __gst_once_init_enter(val) (g_once_init_enter (val)) > > #define __gst_once_init_leave(val,newval) (g_once_init_leave (val, > > newval)) > > #else > > #define __gst_once_init_enter(val) (G_UNLIKELY (*(val) == 0)) > > #define __gst_once_init_leave(val,newval) (*(val) = newval) > > #endif > > " > > > > Any explanations ? > > > > Sincerely > > > > Julien > > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From thaytan at noraisin.net Tue Jan 20 00:59:13 2009 From: thaytan at noraisin.net (thaytan at noraisin.net) Date: Mon, 19 Jan 2009 23:59:13 +0000 (GMT) Subject: [gst-devel] RELEASE: GStreamer 0.10.22 'I heard a disturbing sound' Message-ID: <20090119235913.66CC7C9804D@mediacenter> This mail announces the release of GStreamer 0.10.22 'I heard a disturbing sound'. GStreamer is a streaming media framework that allows the construction of graphs of elements which operate on media data. Applications using this library can do anything from real-time sound processing over playing video to capturing audio, video, and even other types of media data. Its architecture allows for adding new data types or processing capabilities simply by installing new plug-ins. GStreamer is the core module, containing libraries, headers, the basic object hierarchy, and a set of media-agnostic core elements. For more information, see http://gstreamer.freedesktop.org/modules/gstreamer.html To file bugs, go to http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gstreamer+%28core%29 -------------- next part -------------- Release notes for GStreamer??0.10.22 "I heard a disturbing sound" The GStreamer team is proud to announce a new release in the 0.10.x stable series of the core of the GStreamer streaming media framework. The 0.10.x series is a stable series targeted at end users. It is not API or ABI compatible with the stable 0.8.x series. It is, however, parallel installable with the 0.8.x series. The 0.10.x series has been reworked for threadsafety. It also features various feature additions and enhancements. This module, gstreamer, only contains core functionality. For actual media playback, you will need other modules. gst-plugins-base contains a basic set of well-supported plug-ins gst-plugins-good contains a set of well-supported plug-ins under our preferred license gst-plugins-ugly contains a set of well-supported plug-ins, but might pose problems for distributors gst-plugins-bad contains a set of less supported plug-ins that haven't passed the rigorous quality testing we expect Features of this release * GstController improvements * Extensions to the latency setting behaviours * Ability for plugins to register dependencies to trigger re-scanning * Optimisations in some common operations on caps and values and elsewhere * Add sequence numbers to events so dependent events can be associated with the event that caused them * New macros for reading and writing float values * Many improvements in the basesink base class * Support non-default main contexts for GstBus watches * Improvements in pull-mode scheduling of sinks * Fixes for alignment issues on sparc * Ghost pad fixes * New bit-reader assistance API * Many other bug fixes and improvements Bugs fixed in this release * 567002 : --gst-disable-registry-update also skips registry creation * 350477 : [Registry] Provide a way for plugins to delegate the 'cha... * 510354 : post GST_MESSAGE_STRUCTURE_CHANGE when linking and unlink... * 526044 : Crash if there is more than one watcher for a GstBus * 539108 : gst_ghost_pad_new() does more than call g_object_new() * 544003 : [parse] Links to wrong pad if no padtemplate name is given * 544293 : rtspsrc / gstpoll regression * 545714 : GST_READ_UINT_* and GST_WRITE_UINT_* are not always safe ... * 548786 : [win32] search for plugins based on location of DLL * 551676 : Small Memleak fix in gst_pad_link_check_compatible_unlocked * 552778 : add free_func in gstbuffer instances * 553554 : [base] Add a generic bit reader * 553874 : query_position broken during seek * 555196 : Move floating point endian conversion macros into core * 555307 : Need to push latency event into the pipeline * 555631 : Banshee crashes; gstreamer GST_CAPS GST_IS_CAPS warnings * 556091 : libtool-2.2 and old gettext don't mix * 556900 : a copy & paste bug in helloworld example * 557154 : Allow totem to run without write-access to home directory * 557300 : libgstnet needs to link against $(WIN32_LIBS) * 557438 : Gstreamer crashes when Banshee supposed to change track * 557649 : GstBaseTransform can cause ivalid memory references * 559250 : segments need identifiers. * 560301 : Add --plugin to gst-inspect * 560360 : Excessive statting when reading binary cache * 560399 : Unnecessary CRC checksum for binary registry cache * 560442 : GNOME Goal: Remove deprecated GLib symbols * 561501 : g_type_class_ref without unref * 562170 : GstBus watch doesn't work with non-default main context * 562949 : [API] Add an URI query type for querying the source/sink URI * 564863 : Caps are not propagated from a ghostpad's target to the g... * 567225 : volatile gonce_data in BOILERPLATE not initialized to 0 * 566393 : [deadlock] Setting index on a pipeline API changed in this release - API additions: * Add bit reader and byte reader classes. * gst_message_new_structure_change() * gst_message_parse_structure_change() * GST_BUFFER_FREE_FUNC() * Add GST_READ_UINT24_(LE|BE) and GST_WRITE_UINT24_(LE|BE). * GstBaseSrc::gst_base_src_set_blocksize() * GstBaseSrc::gst_base_src_get_blocksize() * GstBaseSink::gst_base_sink_set_blocksize() * GstBaseSink::gst_base_sink_get_blocksize() * GstBaseSink::blocksize * GstBaseSink::gst_base_sink_do_preroll() * Add gst_byte_reader_get_data and gst_byte_reader_peek_data * Move float endianness conversion macros from libgstfloatcast to core * gst_util_seqnum_next, gst_util_seqnum_compare * gst_event_get_seqnum, gst_event_set_seqnum * gst_message_get_seqnum, gst_message_set_seqnum * GstSeekFlags::GST_SEEK_FLAG_SKIP * gst_bin_recalculate_latency() * GstBin::do-latency * gst_tag_setter_reset_tags() * gst_query_new_uri, gst_query_set_uri, gst_query_parse_uri * add gst_plugin_add_dependency() * add gst_plugin_add_dependency_simple() Download You can find source releases of gstreamer in the download directory: http://gstreamer.freedesktop.org/src/gstreamer/ GStreamer Homepage More details can be found on the project's website: http://gstreamer.freedesktop.org/ Support and Bugs We use GNOME's bugzilla for bug reports and feature requests: http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer Developers CVS is hosted on cvs.freedesktop.org. All code is in CVS and can be checked out from there. Interested developers of the core library, plug-ins, and applications should subscribe to the gstreamer-devel list. If there is sufficient interest we will create more lists as necessary. Applications Applications ported to GStreamer 0.10 include Totem, RhythmBox, Sound-Juicer, Gnome Media, Flumotion, Amarok, Jamboree, Pitivi, Istanbul, AnnoAmp, Elisa, and others. Let us know if you want to be added to this list. Contributors to this release * Alessandro Decina * Andrew Feren * Andy Wingo * Antoine Tremblay * Cygwin Ports maintainer * David Schleef * Dig Ge * Edward Hervey * Jan Schmidt * Jan Schmidt - Sun Microsystems * Jonas Holmberg * Michael Smith * Olivier Crete * Peter Kjellerstedt * Sebastian Dr??ge * Simon Holm Th??gersen * Stefan Kost * Thijs Vermeir * Thomas Vander Stichele * Tim-Philipp M??ller * Wim Taymans * ????????? ?? From thaytan at noraisin.net Tue Jan 20 01:00:07 2009 From: thaytan at noraisin.net (thaytan at noraisin.net) Date: Tue, 20 Jan 2009 00:00:07 +0000 (GMT) Subject: [gst-devel] RELEASE: GStreamer Base Plug-ins 0.10.22 'Hidey Hidey Hidey Ho' Message-ID: <20090120000007.8C656C9804D@mediacenter> This mail announces the release of GStreamer Base Plug-ins 0.10.22 'Hidey Hidey Hidey Ho'. GStreamer Base Plug-ins is a well-groomed and well-maintained collection of GStreamer plug-ins and elements, spanning the range of possible types of elements one would want to write for GStreamer. It also contains helper libraries and base classes useful for writing elements. A wide range of video and audio decoders, encoders, and filters are included. For more information, see http://gstreamer.freedesktop.org/modules/gst-plugins-base.html To file bugs, go to http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gst-plugins-base -------------- next part -------------- Release notes for GStreamer Base Plug-ins??0.10.22 "hidey hidey hidey ho" The GStreamer team is proud to announce a new release in the 0.10.x stable series of the GStreamer Base Plug-ins. The 0.10.x series is a stable series targeted at end users. It is not API or ABI compatible with the stable 0.8.x series. It is, however, parallel installable with the 0.8.x series. This module contains a set of reference plugins, base classes for other plugins, and helper libraries. This module is kept up-to-date together with the core developments. Element writers should look at the elements in this module as a reference for their development. This module contains elements for, among others: device plugins: x(v)imagesink, alsa, v4lsrc, cdparanoia containers: ogg codecs: vorbis, theora text: textoverlay, subparse sources: audiotestsrc, videotestsrc, gnomevfssrc network: tcp typefind audio processing: audioconvert, adder, audiorate, audioscale, volume visualisation: libvisual video processing: ffmpegcolorspace aggregate elements: decodebin, playbin Other modules containing plug-ins are: gst-plugins-good contains a set of well-supported plug-ins under our preferred license gst-plugins-ugly contains a set of well-supported plug-ins, but might pose problems for distributors gst-plugins-bad contains a set of less supported plug-ins that haven't passed the rigorous quality testing we expect Features of this release * Require gettext 0.17 * Replace audioresample with speexresample from -bad * Support new formats in RIFF: uncompressed RGB, WMA lossless, VP6 * Move libgstapp and elements from -bad * Support color-key setting and probing for Xv properties * Improve typefinding for various formats * Extend audio sinks for pull-mode operation * Support for more subtitle formats * More development on decode2bin and playbin2 * RTP and SDP fixes * Many bug fixes and improvements Bugs fixed in this release * 562163 : theoraenc likely ignoring segments * 562258 : rtspsrc element takes long time to error out if the addre... * 561789 : [volume] deadlocks with a controller attached * 554533 : [xvimagesink] allow setting colorkey if possible * 567511 : colorkey in xvimagesink gets reset when element is reused * 116051 : libresample doesn't handle > factor of 2 rate conversion * 346218 : [audioresample] doesn't do anti aliasing * 385061 : [audioresample?] investigate high CPU usage * 456788 : [subparse] can't handle UTF-16 charset encoded subtitle. * 525807 : [vorbisenc] vorbisenc has problems with a gnlsource that ... * 546955 : gstoggmux EOS handling issue * 549417 : [audioresample] unit test fails on 64bit linux * 549510 : audioresample doesn't negotiate ideal caps * 552237 : UTF-16 srt confuses gstreamer, misdetected as mp3 * 552559 : Implementation of SLAVE_SKEW in baseaudiosrc * 552569 : audioresample producing strange sized buffers * 552801 : audioconvert can overflow with big audio buffers * 554879 : Add ability to specify format for date/time display in Gs... * 555257 : Doesn't display srt subtitles saved with BOM * 555319 : add FFV1 fourcc to riff-media * 555607 : subrip subtitles typefind too strict * 555699 : [PATCH] theoradec: prefer container's pixel aspect ratio ... * 556025 : build failure in tests/icles * 556066 : Last byte of FLAC image buffer chopped off * 557365 : subparse check fails * 558124 : [PLUGIN-MOVE] Move speexresample as audioresample2 to -base * 559111 : ALSA sink hangs on USB audio device unplug while playing * 559478 : does not play windows media streams correctly * 559567 : `gst_base_audio_sink_sync_latency' should call `gst_base_... * 561436 : videorate element add image/jpeg to caps template * 561734 : playbin2 additions * 561780 : Playbin2 should work without volume too * 561924 : oggdemux hangs when given corrupt input via non-seekable ... * 562270 : build without gdk fails * 563143 : ximagesink/xvimagesink : _alloc_buffer returns non-clean ... * 563174 : Implement gst_rtcp_packet_remove * 563508 : [rgvolume] Unit test fails with passthrough assertions * 563718 : Theora check out of date * 563904 : GNOME Goal: Clean up GLib and GTK+ includes * 564098 : MS Word files are recognised as audio/mpeg and OSX's .DS_... * 564139 : Documentation of TCP plugins * 564200 : GstBaseAudioSink should register its enums and have corre... * 564206 : GstBaseAudioSrc should register its enum and have corresp... * 564421 : Move appsrc/appsink to -base * 564929 : Audiosink blocks if setcaps called while playing * 566586 : playbin2 test7.c fails after two songs * 566750 : [appsrc/sink] add padding, move private data to private s... * 566761 : [gstapp] No pkg-config file * 566837 : gst_cdda_base_src_mode_get_type() is not public from < gst... * 566875 : [gnomevfs] Add dependency for the GnomeVFS modules * 566876 : [gio] Add dependency for the modules dir * 567027 : Add GType for GstRTSPUrl for bindings * 567168 : appsink is using the wrong signal slot for the pull-buffe... * 567960 : [tagdemux] Doesn't forward unknown events upstream * 500833 : [FFT] Struct alignment issues on sparc * 552199 : Parsing SDP file with multicast address fails * 558553 : [riff] gst_riff_create_video_caps not recognizing certain... * 564896 : gst_netaddress_get_ip[46]_address should check for correc... * 566341 : Some Ogg Theora files don't finished at seek at the end * 566654 : playbin2: does not come back from NULL after switching UR... * 566723 : GstAudioClock's new function may better use const gchar* ... API changed in this release - API additions: * clockoverlay::time-format * GstRingBuffer:gst_ring_buffer_activate() * GstRingBuffer:gst_ring_buffer_is_active() * GstRingBuffer:gst_ring_buffer_convert() * Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API * gst_netaddress_get_address_bytes() * gst_netaddress_set_address_bytes() Download You can find source releases of gst-plugins-base in the download directory: http://gstreamer.freedesktop.org/src/gst-plugins-base/ GStreamer Homepage More details can be found on the project's website: http://gstreamer.freedesktop.org/ Support and Bugs We use GNOME's bugzilla for bug reports and feature requests: http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer Developers CVS is hosted on cvs.freedesktop.org. All code is in CVS and can be checked out from there. Interested developers of the core library, plug-ins, and applications should subscribe to the gstreamer-devel list. If there is sufficient interest we will create more lists as necessary. Applications Contributors to this release * Alessandro Decina * Andrew Feren * Andy Wingo * Christian Schaller * Cygwin Ports maintainer * Damien Lespiau * Daniel Drake * David Schleef * Edward Hervey * Guillaume Emont * H??vard Graff * Jan Gerber * Jan Schmidt * Jonathan Matthew * Jonathan Rosser * Jos?? Alburquerque * Julien Moutte * Klaas * Luis Menina * Mark Nauwelaerts * Matthias Kretz * Michael Smith * Nick Haddad * Olivier Crete * Pavel Zeldin * Robin Stocker * Sebastian Dr??ge * Stefan Kost * Tero Saarni * Thomas Vander Stichele * Tim-Philipp M??ller * Wim Taymans * xavierb at gmail dot com * ????????? ?? From thaytan at noraisin.net Tue Jan 20 01:01:22 2009 From: thaytan at noraisin.net (thaytan at noraisin.net) Date: Tue, 20 Jan 2009 00:01:22 +0000 (GMT) Subject: [gst-devel] RELEASE: GStreamer Python Bindings 0.10.14 'You Better Think' Message-ID: <20090120000122.D1086C98050@mediacenter> This mail announces the release of GStreamer Python Bindings 0.10.14 'You Better Think'. GStreamer Python Bindings is a set of Python bindings for GStreamer, using the same system as PyGTK. It also comes with a number of examples. For more information, see http://gstreamer.freedesktop.org/modules/gst-python.html To file bugs, go to http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gst-python -------------- next part -------------- Release notes for GStreamer Python bindings??0.10.14 "you better think" The GStreamer team is proud to announce a new release in the 0.10.x stable series of the GStreamer Python Bindings. The 0.10.x series is a stable series targeted at end users. It is not API or ABI compatible with the stable 0.8.x series. It is, however, parallel installable with the 0.8.x series. Features of this release * Handle keep-refcount for GBoxed arguments * Mark gst.message* that take ownership of the gst.Structure/gst.TagList * Wrap new API in Core and Base * Various other bug fixes and improvements Bugs fixed in this release * 543961 : Memory leak in gst.Bin.do_handle_message * 553131 : Import modules in correct order * 553134 : Assumes wrong RTLD constants for Linux/MIPS * 554545 : gst.Pad methods that return a new buffer as function argu... * 556054 : codegen: Add 'keep-refcount' handling for GBoxed arguments * 559406 : Add proxy for gst_base_transform_transform_size * 559768 : Fix gst-python on systems that don't support RTLD_GLOBAL ... * 564580 : Memory leak in gst.Structure.keys() * 566903 : gst.Pad.iterate_internal_links may raise critical warning... Download You can find source releases of gst-python in the download directory: http://gstreamer.freedesktop.org/src/gst-python/ GStreamer Homepage More details can be found on the project's website: http://gstreamer.freedesktop.org/ Support and Bugs We use GNOME's bugzilla for bug reports and feature requests: http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer Developers CVS is hosted on cvs.freedesktop.org. All code is in CVS and can be checked out from there. Interested developers of the core library, plug-ins, and applications should subscribe to the gstreamer-devel list. If there is sufficient interest we will create more lists as necessary. Applications Contributors to this release * Alessandro Decina * Alexander Wirt * Edward Hervey * Jan Schmidt * Sebastian Dr??ge * Thiemo Seufer * Vincent GENIEUX ?? From thaytan at noraisin.net Tue Jan 20 01:01:10 2009 From: thaytan at noraisin.net (thaytan at noraisin.net) Date: Tue, 20 Jan 2009 00:01:10 +0000 (GMT) Subject: [gst-devel] RELEASE: GStreamer Bad Plug-ins 0.10.10 'Keep the dogies rollin' Message-ID: <20090120000110.C6D40C9804D@mediacenter> This mail announces the release of GStreamer Bad Plug-ins 0.10.10 'Keep the dogies rollin'. GStreamer Bad Plug-ins is a set of plug-ins that aren't up to par compared to the rest. They might be close to being good quality, but they're missing something - be it a good code review, some documentation, a set of tests, a real live maintainer, or some actual wide use. For more information, see http://gstreamer.freedesktop.org/modules/gst-plugins-bad.html To file bugs, go to http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gst-plugins-bad -------------- next part -------------- Release notes for GStreamer Bad Plug-ins??0.10.10 "keep them dogies rollin" The GStreamer team is proud to announce a new release in the 0.10.x stable series of the GStreamer Bad Plug-ins. The 0.10.x series is a stable series targeted at end users. It is not API or ABI compatible with the stable 0.8.x series. It is, however, parallel installable with the 0.8.x series. "That an accusation?" No perfectly groomed moustache or any amount of fine clothing is going to cover up the truth - these plug-ins are Bad with a capital B. They look fine on the outside, and might even appear to get the job done, but at the end of the day they're a black sheep. Without a golden-haired angel to watch over them, they'll probably land in an unmarked grave at the final showdown. Don't bug us about their quality - exercise your Free Software rights, patch up the offender and send us the patch on the fastest steed you can steal from the Confederates. Because you see, in this world, there's two kinds of people, my friend: those with loaded guns and those who dig. You dig. This module contains a set of plug-ins that aren't up to par compared to the rest. They might be close to being good quality, but they're missing something - be it a good code review, some documentation, a set of tests, a real live maintainer, or some actual wide use. If the blanks are filled in they might be upgraded to become part of either gst-plugins-good or gst-plugins-ugly, depending on the other factors. If the plug-ins break, you can't complain - instead, you can fix the problem and send us a patch, or bribe someone into fixing them for you. New contributors can start here for things to work on. Other modules containing plug-ins are: gst-plugins-base contains a basic set of well-supported plug-ins gst-plugins-good contains a set of well-supported plug-ins under our preferred license gst-plugins-ugly contains a set of well-supported plug-ins, but might pose problems for distributors Features of this release * Require gettext 0.17 * ACM mp3 decoder element for win32 * New MXF demuxer added * JPEG-2000 encoder - jp2enc added * RTP support * Fixes for RTP support * Add parsers for AAC and AMR * Move libgstapp and elements to -base * Add Quicktime muxer element qtmux * Enhance RFB capture * Improve MPEG-TS and MPEG-PS demuxing * Don't install static libs for plugins * Introduce audioringbuffer element * Improve DVD playback of ResinDVD components * Move old audioresample from -base as legacyaudioresample * Improve speexresampler, and then move it to -base * Improve FLV demuxing and parsing * Various DVB input fixes * Add support for frequency list descriptors for DVB * Improve MPEG-TS muxer * Various other bug-fixes Bugs fixed in this release * 127574 : replaygain support * 480543 : [spc] crashes when not given any input data * 546319 : [resindvd] Stuck on first menu (language selection) * 549222 : Totem using resindvd element as DVD navigation module doe... * 550280 : Move qtmux to gst-plugins-bad * 550454 : FLV demuxer screws up in Pitivi * 550613 : [mpegtsmux] Add support for ac3/dts/lpcm * 553823 : CodeStyle improvements and minor fixes to the DCCP plugin. * 555238 : Recent changes in x264 breaks compilation of x264 plugin. * 557161 : Wrong time displayed and no seeking with MPEG PS cut from... * 557763 : [mpegtsmux] DeadlockEOS due to interleaving mechanism * 557814 : [mpegdemux] frequency list descriptor support * 558639 : cross-configuring interrupts due to a failing check for " ... * 560841 : Provide access to packet loss * 561992 : debug format warning during compilation of resindvdsrc.c ... * 562312 : Lip-sync generates wrong `ts-offset' in GstRtpBin * 562319 : [rtpsession] memory corruption * 562802 : [appsrc] lockup in pad_push() if previous flow was wrong-... * 563006 : mxf mpeg422 file stops playing * 563147 : input-selector doesn't notify of active-pad changing when... * 564232 : Dual Licensing qtmux * 565439 : Make pcapparse compile with (mingw32) gcc * 565509 : rtpsource on receiver side can't get clock-rate for h264,... * 565614 : [mpegtsdemux] Fix memory leak * 565910 : allow forcing the ssrc on a rtpsession * 566356 : [dvbbasebin] Couple of memleaks * 566613 : bayer2rgb calculates wrong color offset * 566878 : [ladspa] Add dependencies for ladspa modules * 566957 : Add time based seeking to ResinDVD * 566958 : fix dirac support in qtmux/mp4mux * 566959 : Fix duration query on mpegtsdemux * 508029 : [speexresample] pops/glitches in some cases * 558416 : [PATCH] realvideodec: use pixel aspect ratio of container * 564482 : [appsrc] semantics of push-buffer signal awkward for bind... * 565319 : rtpjitterbuffer: bursty reception can cause time to be re... * 565613 : [mpegtsdemux] Add scrambled field to service structure * 567004 : ladspa plugin crashes on Solaris Download You can find source releases of gst-plugins-bad in the download directory: http://gstreamer.freedesktop.org/src/gst-plugins-bad/ GStreamer Homepage More details can be found on the project's website: http://gstreamer.freedesktop.org/ Support and Bugs We use GNOME's bugzilla for bug reports and feature requests: http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer Developers CVS is hosted on cvs.freedesktop.org. All code is in CVS and can be checked out from there. Interested developers of the core library, plug-ins, and applications should subscribe to the gstreamer-devel list. If there is sufficient interest we will create more lists as necessary. Applications Contributors to this release * Alessandro Decina * Arnout Vandecappelle * Brian Cameron * Christian Schaller * Cygwin Ports maintainer * Damien Lespiau * David H??rdeman * David Schleef * Edward Hervey * Jan Schmidt * Jonathan Matthew * Josep Torra * Julien Moutte * Leandro Melo de Sales * Luotao Fu * Mark Nauwelaerts * Michael Smith * Olivier Crete * Robin Stocker * Sascha Hauer * Sebastian Dr??ge * Sebastian P??lsterl * Stefan Kost * Tal Shalif * Thiago Sousa Santos * Thijs Vermeir * Thomas Vander Stichele * Tim-Philipp M??ller * Wim Taymans * Zaheer Abbas Merali * vanista * ????????? ?? From thaytan at noraisin.net Tue Jan 20 01:20:08 2009 From: thaytan at noraisin.net (Jan Schmidt) Date: Tue, 20 Jan 2009 00:20:08 +0000 Subject: [gst-devel] Core/Base/Bad/Python released. Next step - moving to GIT. Message-ID: <1232410808.1311.20.camel@fancy-ubuntu> Now that the set of Core/Base/Python/Bad release are out of the way, the plan is to move to GIT over the next week. Edward will be managing the timing and activity of the actual migration to fit in with his week at LCA, but the general plan is: * Core/Base/Bad/Python stay frozen until further notice * Edward decides when he wants to cue up the migration and lets us know * At a designated time *all other* modules will freeze in CVS and CVS will be marked Read-only, so that there can be no commits while the migration takes place. * Everything moves to git in the standard locations on the freedesktop.org servers (which should be something like http://cgit.freedesktop.org/gstreamer/gstreamer, http://cgit.freedesktop.org/gstreamer/gst-plugins-base etc) * Everyone can start using git, and CVS is only useful as a read-only history. In case of unexpected disaster: All repositories are tagged with GIT_MIGRATION for the commit at which the conversions were done. If the worst comes to worst, we can then easily get all commits since that conversion as a series of unified-diff patches, make cvs read-write again, and commit them. Edward - over to you! Cheers, Jan. -- Jan Schmidt From bilboed at gmail.com Tue Jan 20 04:23:58 2009 From: bilboed at gmail.com (Edward Hervey) Date: Tue, 20 Jan 2009 04:23:58 +0100 Subject: [gst-devel] Core/Base/Bad/Python released. Next step - moving to GIT. In-Reply-To: <1232410808.1311.20.camel@fancy-ubuntu> References: <1232410808.1311.20.camel@fancy-ubuntu> Message-ID: <1232421838.25255.6.camel@joder> Thanks Jan for the releases ! So for the missing info: * I'll be grabbing an fdo admin to de-activate cvs (leaving only anoncvs). No shortage of them here at LCA. * Since this switch information has already been circulating for a few weeks now, I propose to do that at 23:59 CET tonight (20th Jan). For people in Europe it won't matter that way, but I'm currently in Australia so leaves me a full day to make sure the migration goes flawlessly before Europe wakes up again. The rest is pretty straighforward, and the conversion/upload/activate should take (with checks) around 4 hours. I don't think there's much more to say, See you the other side :) Edward On Tue, 2009-01-20 at 00:20 +0000, Jan Schmidt wrote: > Now that the set of Core/Base/Python/Bad release are out of the way, the > plan is to move to GIT over the next week. > > Edward will be managing the timing and activity of the actual migration > to fit in with his week at LCA, but the general plan is: > > * Core/Base/Bad/Python stay frozen until further notice > * Edward decides when he wants to cue up the migration and lets us know > * At a designated time *all other* modules will freeze in CVS and CVS > will be marked Read-only, so that there can be no commits while the > migration takes place. > * Everything moves to git in the standard locations on the > freedesktop.org servers (which should be something like > http://cgit.freedesktop.org/gstreamer/gstreamer, > http://cgit.freedesktop.org/gstreamer/gst-plugins-base etc) > * Everyone can start using git, and CVS is only useful as a read-only > history. > > In case of unexpected disaster: > All repositories are tagged with GIT_MIGRATION for the commit at which > the conversions were done. If the worst comes to worst, we can then > easily get all commits since that conversion as a series of unified-diff > patches, make cvs read-write again, and commit them. > > Edward - over to you! > > Cheers, > Jan. From vaisakh.n at nestgroup.net Tue Jan 20 06:43:13 2009 From: vaisakh.n at nestgroup.net (vaisakh.n) Date: Tue, 20 Jan 2009 11:13:13 +0530 Subject: [gst-devel] add a plugin Message-ID: <49756471.60105@nestgroup.net> Hi, I have a decoder plugin's library file ( .so file ) how can i add it to gstreamer plugin registry.. please help me what are the steps i need to follow? advance thanks ***** Confidentiality Statement/Disclaimer ***** This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. From rakesh2.sharma at aricent.com Tue Jan 20 06:48:18 2009 From: rakesh2.sharma at aricent.com (rakesh sharma) Date: Mon, 19 Jan 2009 21:48:18 -0800 (PST) Subject: [gst-devel] How to register Gstreamer Plugin with Playbin so that playbin detect my plugins In-Reply-To: <3c1737210901191016s1848ee38q99b52ea8cec1bb2@mail.gmail.com> References: <21541362.post@talk.nabble.com> <3c1737210901191016s1848ee38q99b52ea8cec1bb2@mail.gmail.com> Message-ID: <21557477.post@talk.nabble.com> I have written mp4sp decoder plugin and i want this plugin should be detected and used by playbin .... Can anybody help me with a simple example....!!! Michael Smith-7 wrote: > > On Mon, Jan 19, 2009 at 3:35 AM, rakesh sharma > wrote: >> >> Hi Friends I have written one plugin now i want that the plugin should be >> called with playbin.... >> can anybody please help me its really urgent...!!!! > > You haven't given much context for us to go on here. > > Is your plugin a source? A sink? A demuxer or decoder? > > The key points to check, if you want playbin to autoplug it, are: > - correct element details class > - non-zero rank (and higher than other alternatives playbin might select) > - correct caps templates. > > We can't be any more specific with the info you've given. > > Mike > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- View this message in context: http://www.nabble.com/How-to-register-Gstreamer-Plugin-with-Playbin-so-that-playbin-detect-my-plugins-tp21541362p21557477.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From bisht.sudarshan at gmail.com Tue Jan 20 06:58:10 2009 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Tue, 20 Jan 2009 11:28:10 +0530 Subject: [gst-devel] add a plugin In-Reply-To: <49756471.60105@nestgroup.net> References: <49756471.60105@nestgroup.net> Message-ID: <785339900901192158o7ffe4087vd2e54e8177c90661@mail.gmail.com> Hi , Set GST_PLUGIN_PATH={ path of .so file } On Tue, Jan 20, 2009 at 11:13 AM, vaisakh.n wrote: > Hi, > I have a decoder plugin's library file ( .so file ) > how can i add it to gstreamer plugin registry.. > please help me > what are the steps i need to follow? > advance thanks > > ***** Confidentiality Statement/Disclaimer ***** > > This message and any attachments is intended for the sole use of the > intended recipient. It may contain confidential information. Any > unauthorized use, dissemination or modification is strictly prohibited. If > you are not the intended recipient, please notify the sender immediately > then delete it from all your systems, and do not copy, use or print. > Internet communications are not secure and it is the responsibility of the > recipient to make sure that it is virus/malicious code exempt. > The company/sender cannot be responsible for any unauthorized alterations > or modifications made to the contents. If you require any form of > confirmation of the contents, please contact the company/sender. The > company/sender is not liable for any errors or omissions in the content of > this message. > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From vacek008 at umn.edu Tue Jan 20 06:59:20 2009 From: vacek008 at umn.edu (Tom Vacek) Date: Mon, 19 Jan 2009 23:59:20 -0600 Subject: [gst-devel] add a plugin In-Reply-To: <49756471.60105@nestgroup.net> References: <49756471.60105@nestgroup.net> Message-ID: <49756838.7080209@umn.edu> Set the variable GST_PLUGIN_PATH=path_to_your_plugin's_directory Then use gst-inspect to make sure it's working. To add the plugin permanently, find your gst plugins directory, and put the library there. It's probably at /usr/lib/gstreamer-0.10/ vaisakh.n wrote: > Hi, > I have a decoder plugin's library file ( .so file ) > how can i add it to gstreamer plugin registry.. > please help me > what are the steps i need to follow? > advance thanks > > ***** Confidentiality Statement/Disclaimer ***** > > This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. > The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From saicharmboy at yahoo.co.in Tue Jan 20 07:06:35 2009 From: saicharmboy at yahoo.co.in (sai charmboy) Date: Tue, 20 Jan 2009 11:36:35 +0530 (IST) Subject: [gst-devel] add a plugin References: <49756471.60105@nestgroup.net> Message-ID: <818500.46623.qm@web94809.mail.in2.yahoo.com> go to hell. dont send messages again saicharmboy... ________________________________ From: vaisakh.n To: Discussion of the development of GStreamer Sent: Tuesday, 20 January, 2009 11:13:13 AM Subject: [gst-devel] add a plugin Hi, I have a decoder plugin's library file ( ..so file ) how can i add it to gstreamer plugin registry.. please help me what are the steps i need to follow? advance thanks ***** Confidentiality Statement/Disclaimer ***** This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. ------------------------------------------------------------------------------ This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From saicharmboy at yahoo.co.in Tue Jan 20 07:07:42 2009 From: saicharmboy at yahoo.co.in (sai charmboy) Date: Tue, 20 Jan 2009 11:37:42 +0530 (IST) Subject: [gst-devel] add a plugin References: <49756471.60105@nestgroup.net> <785339900901192158o7ffe4087vd2e54e8177c90661@mail.gmail.com> Message-ID: <388129.30149.qm@web94815.mail.in2.yahoo.com> go to hell.dont send any messages again.do u understand saicharmboy.... ________________________________ From: sudarshan bisht To: Discussion of the development of GStreamer Sent: Tuesday, 20 January, 2009 11:28:10 AM Subject: Re: [gst-devel] add a plugin Hi , Set GST_PLUGIN_PATH={ path of .so file } On Tue, Jan 20, 2009 at 11:13 AM, vaisakh.n wrote: Hi, I have a decoder plugin's library file ( .so file ) how can i add it to gstreamer plugin registry.. please help me what are the steps i need to follow? advance thanks ***** Confidentiality Statement/Disclaimer ***** This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. ------------------------------------------------------------------------------ This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -- Regards, Sudarshan Bisht Check out the all-new face of Yahoo! India. Go to http://in.yahoo.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From bisht.sudarshan at gmail.com Tue Jan 20 07:09:36 2009 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Tue, 20 Jan 2009 11:39:36 +0530 Subject: [gst-devel] How to register Gstreamer Plugin with Playbin so that playbin detect my plugins In-Reply-To: <21557477.post@talk.nabble.com> References: <21541362.post@talk.nabble.com> <3c1737210901191016s1848ee38q99b52ea8cec1bb2@mail.gmail.com> <21557477.post@talk.nabble.com> Message-ID: <785339900901192209u5b6b87b8h8aa55c78c73947bc@mail.gmail.com> set rank GST_RANK_PRIMARY in plugins_init function. On Tue, Jan 20, 2009 at 11:18 AM, rakesh sharma wrote: > > I have written mp4sp decoder plugin and i want this plugin should be > detected > and used by playbin .... > Can anybody help me with a simple example....!!! > > > > Michael Smith-7 wrote: > > > > On Mon, Jan 19, 2009 at 3:35 AM, rakesh sharma > > wrote: > >> > >> Hi Friends I have written one plugin now i want that the plugin should > be > >> called with playbin.... > >> can anybody please help me its really urgent...!!!! > > > > You haven't given much context for us to go on here. > > > > Is your plugin a source? A sink? A demuxer or decoder? > > > > The key points to check, if you want playbin to autoplug it, are: > > - correct element details class > > - non-zero rank (and higher than other alternatives playbin might > select) > > - correct caps templates. > > > > We can't be any more specific with the info you've given. > > > > Mike > > > > > ------------------------------------------------------------------------------ > > This SF.net email is sponsored by: > > SourcForge Community > > SourceForge wants to tell your story. > > http://p.sf.net/sfu/sf-spreadtheword > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > -- > View this message in context: > http://www.nabble.com/How-to-register-Gstreamer-Plugin-with-Playbin-so-that-playbin-detect-my-plugins-tp21541362p21557477.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From vaisakh.n at nestgroup.net Tue Jan 20 07:09:36 2009 From: vaisakh.n at nestgroup.net (vaisakh.n) Date: Tue, 20 Jan 2009 11:39:36 +0530 Subject: [gst-devel] add a plugin In-Reply-To: <785339900901192158o7ffe4087vd2e54e8177c90661@mail.gmail.com> References: <49756471.60105@nestgroup.net> <785339900901192158o7ffe4087vd2e54e8177c90661@mail.gmail.com> Message-ID: <49756AA0.8090909@nestgroup.net> sudarshan bisht wrote: > Hi , > > Set GST_PLUGIN_PATH={ path of .so file } > > > On Tue, Jan 20, 2009 at 11:13 AM, vaisakh.n > wrote: > > Hi, > I have a decoder plugin's library file ( .so file ) > how can i add it to gstreamer plugin registry.. > please help me > what are the steps i need to follow? > advance thanks > > ***** Confidentiality Statement/Disclaimer ***** > > This message and any attachments is intended for the sole use of > the intended recipient. It may contain confidential information. > Any unauthorized use, dissemination or modification is strictly > prohibited. If you are not the intended recipient, please notify > the sender immediately then delete it from all your systems, and > do not copy, use or print. Internet communications are not secure > and it is the responsibility of the recipient to make sure that it > is virus/malicious code exempt. > The company/sender cannot be responsible for any unauthorized > alterations or modifications made to the contents. If you require > any form of confirmation of the contents, please contact the > company/sender. The company/sender is not liable for any errors or > omissions in the content of this message. > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > -- > Regards, > > Sudarshan Bisht > >------------------------------------------------------------------------ > >------------------------------------------------------------------------------ >This SF.net email is sponsored by: >SourcForge Community >SourceForge wants to tell your story. >http://p.sf.net/sfu/sf-spreadtheword > >------------------------------------------------------------------------ > >_______________________________________________ >gstreamer-devel mailing list >gstreamer-devel at lists.sourceforge.net >https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > Hi, no its not working plugin is written by us .its does not have any api to link with gstreamer.. ***** Confidentiality Statement/Disclaimer ***** This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bisht.sudarshan at gmail.com Tue Jan 20 07:18:34 2009 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Tue, 20 Jan 2009 11:48:34 +0530 Subject: [gst-devel] add a plugin In-Reply-To: <49756AA0.8090909@nestgroup.net> References: <49756471.60105@nestgroup.net> <785339900901192158o7ffe4087vd2e54e8177c90661@mail.gmail.com> <49756AA0.8090909@nestgroup.net> Message-ID: <785339900901192218r5be57781q1c4b93ec0f003935@mail.gmail.com> You might be having plugin_init function in your plugin code where u give name of your plugin right ? On Tue, Jan 20, 2009 at 11:39 AM, vaisakh.n wrote: > sudarshan bisht wrote: > > Hi , > > Set GST_PLUGIN_PATH={ path of .so file } > > > On Tue, Jan 20, 2009 at 11:13 AM, vaisakh.n wrote: > >> Hi, >> I have a decoder plugin's library file ( .so file ) >> how can i add it to gstreamer plugin registry.. >> please help me >> what are the steps i need to follow? >> advance thanks >> >> ***** Confidentiality Statement/Disclaimer ***** >> >> This message and any attachments is intended for the sole use of the >> intended recipient. It may contain confidential information. Any >> unauthorized use, dissemination or modification is strictly prohibited. If >> you are not the intended recipient, please notify the sender immediately >> then delete it from all your systems, and do not copy, use or print. >> Internet communications are not secure and it is the responsibility of the >> recipient to make sure that it is virus/malicious code exempt. >> The company/sender cannot be responsible for any unauthorized alterations >> or modifications made to the contents. If you require any form of >> confirmation of the contents, please contact the company/sender. The >> company/sender is not liable for any errors or omissions in the content of >> this message. >> >> >> ------------------------------------------------------------------------------ >> This SF.net email is sponsored by: >> SourcForge Community >> SourceForge wants to tell your story. >> http://p.sf.net/sfu/sf-spreadtheword >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > > > -- > Regards, > > Sudarshan Bisht > > ------------------------------ > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story.http://p.sf.net/sfu/sf-spreadtheword > > ------------------------------ > > _______________________________________________ > gstreamer-devel mailing listgstreamer-devel at lists.sourceforge.nethttps://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > Hi, > no its not working > plugin is written by us .its does not have any api to link with > gstreamer.. > > ***** Confidentiality Statement/Disclaimer ***** > > This message and any attachments is intended for the sole use of the > intended recipient. It may contain confidential information. Any > unauthorized use, dissemination or modification is strictly prohibited. If > you are not the intended recipient, please notify the sender immediately > then delete it from all your systems, and do not copy, use or print. > Internet communications are not secure and it is the responsibility of the > recipient to make sure that it is virus/malicious code exempt. > The company/sender cannot be responsible for any unauthorized alterations > or modifications made to the contents. If you require any form of > confirmation of the contents, please contact the company/sender. The > company/sender is not liable for any errors or omissions in the content of > this message. > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From irfanshaikh at tataelxsi.co.in Tue Jan 20 07:39:57 2009 From: irfanshaikh at tataelxsi.co.in (Irfan Shaikh) Date: Tue, 20 Jan 2009 12:09:57 +0530 Subject: [gst-devel] Sync Issue Message-ID: <9D5E1752379A43408015F7FE98466115048C0A69@CHNEXVS01.VSNLXCHANGE.COM> Hi Developers, I have two separate A/V pipelines running at same time, Ex: 1) Pipeline A with elements {a, b, c, d, e} 2) Pipeline B with elements {w, x, y, z} Can i inform a message or signal from element "w" to element "a" by any means. Regards This message (including any attachment) is confidential and may be legally privileged. Access to this message by anyone other than the intended recipient(s) listed above is unauthorized. If you are not the intended recipient you are hereby notified that any disclosure, copying, or distribution of the message, or any action taken or omission of action by you in reliance upon it, is prohibited and may be unlawful. Please immediately notify the sender by reply e-mail and permanently delete all copies of the message if you have received this message in error. -------------- next part -------------- An HTML attachment was scrubbed... URL: From rakesh2.sharma at aricent.com Tue Jan 20 07:50:27 2009 From: rakesh2.sharma at aricent.com (rakesh sharma) Date: Mon, 19 Jan 2009 22:50:27 -0800 (PST) Subject: [gst-devel] How to register Gstreamer Plugin with Playbin so that playbin detect my plugins In-Reply-To: <785339900901192209u5b6b87b8h8aa55c78c73947bc@mail.gmail.com> References: <21541362.post@talk.nabble.com> <3c1737210901191016s1848ee38q99b52ea8cec1bb2@mail.gmail.com> <21557477.post@talk.nabble.com> <785339900901192209u5b6b87b8h8aa55c78c73947bc@mail.gmail.com> Message-ID: <21557989.post@talk.nabble.com> This is not correct..... it dosn't have any effect on playbin ...GST_RANK_PRIMARY sudarshan bisht wrote: > > set rank GST_RANK_PRIMARY in plugins_init function. > > On Tue, Jan 20, 2009 at 11:18 AM, rakesh sharma > wrote: > >> >> I have written mp4sp decoder plugin and i want this plugin should be >> detected >> and used by playbin .... >> Can anybody help me with a simple example....!!! >> >> >> >> Michael Smith-7 wrote: >> > >> > On Mon, Jan 19, 2009 at 3:35 AM, rakesh sharma >> > wrote: >> >> >> >> Hi Friends I have written one plugin now i want that the plugin should >> be >> >> called with playbin.... >> >> can anybody please help me its really urgent...!!!! >> > >> > You haven't given much context for us to go on here. >> > >> > Is your plugin a source? A sink? A demuxer or decoder? >> > >> > The key points to check, if you want playbin to autoplug it, are: >> > - correct element details class >> > - non-zero rank (and higher than other alternatives playbin might >> select) >> > - correct caps templates. >> > >> > We can't be any more specific with the info you've given. >> > >> > Mike >> > >> > >> ------------------------------------------------------------------------------ >> > This SF.net email is sponsored by: >> > SourcForge Community >> > SourceForge wants to tell your story. >> > http://p.sf.net/sfu/sf-spreadtheword >> > _______________________________________________ >> > gstreamer-devel mailing list >> > gstreamer-devel at lists.sourceforge.net >> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > >> > >> >> -- >> View this message in context: >> http://www.nabble.com/How-to-register-Gstreamer-Plugin-with-Playbin-so-that-playbin-detect-my-plugins-tp21541362p21557477.html >> Sent from the GStreamer-devel mailing list archive at Nabble.com. >> >> >> >> ------------------------------------------------------------------------------ >> This SF.net email is sponsored by: >> SourcForge Community >> SourceForge wants to tell your story. >> http://p.sf.net/sfu/sf-spreadtheword >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > > > -- > Regards, > > Sudarshan Bisht > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- View this message in context: http://www.nabble.com/How-to-register-Gstreamer-Plugin-with-Playbin-so-that-playbin-detect-my-plugins-tp21541362p21557989.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From luca.ognibene at gmail.com Tue Jan 20 09:44:50 2009 From: luca.ognibene at gmail.com (Luca Ognibene) Date: Tue, 20 Jan 2009 09:44:50 +0100 Subject: [gst-devel] info about dynamic request pads In-Reply-To: <2dc5f8a50901190210g7cf16b3fndcae1c2d4cf206f1@mail.gmail.com> References: <2dc5f8a50901190210g7cf16b3fndcae1c2d4cf206f1@mail.gmail.com> Message-ID: <2dc5f8a50901200044qcc3d9e5uffdd634f0e053d2c@mail.gmail.com> More info of a segfault: * pipeline: http://people.freedesktop.org/~skaboy/76.svg * backtrace: http://people.freedesktop.org/~skaboy/76.bt * GST_DEBUG info: http://people.freedesktop.org/~skaboy/76.log.bz2 Please tell me if you need more info. Luca On Mon, Jan 19, 2009 at 11:10 AM, Luca Ognibene wrote: > Hi, i would like to create a bin that works this way: > * a single sink pad > * src%d are request pad > * inside the bin i have the sink pad linked to a tee > * every time i request a pad i want my bin to create a "queue ! > identity" bin and attach it to the internal tee > * every time i release a pad i want my bin to stop the "queue ! > identity" bin and release the internal tee pad > > Everything work if the bin is in the NULL state but i have > segfaults/hangs if the bin is in PLAYING state. I've tried to read > part-block and other > docs but i'm not able to fix this problem.. > > I've implemented the bin this way: > * in _init the internal tee is created > > * test app call bin.get_request_pad('src0') > > * in _request_new_pad: > * get element state > * request a pad from internal tee > * block this pad if state is PLAYING > * add the "queue ! identity" bin > * create the ghost pad and set target to bin.src > * link tee request pad to bin.sink > * call sync_state_with_parent on the bin (is this ok? right now > bin.src is not linked and the bin is in playing state) > * unblock tee request pad > > * test app block received ghost pad > * add queue ! fakesink > * link them to ghost pad > * call sync_state_with_parent on queue and fakesink > * unblock ghost pad > > Problems: > * sometime (really often) program hangs on blocking ghost pad > * sometime (not very often) i have segfaults while queue is trying to > fixate caps > > A strange thing is that if i change my bin to add "identity ! > identity" instead of "queue ! identity" everything works fine.. but i > don't know why! And i need the queue there :) > > This is a svg graph of my pipeline after having received the ghost > pad: http://people.freedesktop.org/~skaboy/test76-3.dot.svg > I'm using gstreamer 0.10.21 (core). > > ciao > Luca > From t.i.m at zen.co.uk Tue Jan 20 10:24:40 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Tue, 20 Jan 2009 09:24:40 +0000 Subject: [gst-devel] How to register Gstreamer Plugin with Playbin so that playbin detect my plugins In-Reply-To: <21557477.post@talk.nabble.com> References: <21541362.post@talk.nabble.com> <3c1737210901191016s1848ee38q99b52ea8cec1bb2@mail.gmail.com> <21557477.post@talk.nabble.com> Message-ID: <1232443480.487.6.camel@mini.centricular.net> On Mon, 2009-01-19 at 21:48 -0800, rakesh sharma wrote: > I have written mp4sp decoder plugin and i want this plugin should be detected > and used by playbin .... Mike has given you all the details you need. How about you give us the gst-inspect output for your element so we can double-check the details? Alternatively, you could provide us with details on *all* of the points Mike mentioned, namely: > > The key points to check, if you want playbin to autoplug it, are: > > - correct element details class > > - non-zero rank (and higher than other alternatives playbin might select) > > - correct caps templates. > Can anybody help me with a simple example....!!! Almost every single decoder and demuxer in gst-plugins-base and gst-plugins-good will do as an example. Cheers -Tim From t.i.m at zen.co.uk Tue Jan 20 10:27:55 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Tue, 20 Jan 2009 09:27:55 +0000 Subject: [gst-devel] Sync Issue In-Reply-To: <9D5E1752379A43408015F7FE98466115048C0A69@CHNEXVS01.VSNLXCHANGE.COM> References: <9D5E1752379A43408015F7FE98466115048C0A69@CHNEXVS01.VSNLXCHANGE.COM> Message-ID: <1232443675.487.10.camel@mini.centricular.net> On Tue, 2009-01-20 at 12:09 +0530, Irfan Shaikh wrote: > I have two separate A/V pipelines running at same time, > Ex: > 1) Pipeline A with elements {a, b, c, d, e} > 2) Pipeline B with elements {w, x, y, z} > > Can i inform a message or signal from element "w" to element "a" by > any means. It's hard to make any suggestions without more details: - what's the big picture? - what kind of pipeline? - what kind of elements are a + w? - why do they need to communicate directly? - why isn't everything running within the same pipeline? Cheers -Tim From rakesh2.sharma at aricent.com Tue Jan 20 10:50:15 2009 From: rakesh2.sharma at aricent.com (rakesh sharma) Date: Tue, 20 Jan 2009 01:50:15 -0800 (PST) Subject: [gst-devel] How to register Gstreamer Plugin with Playbin so that playbin detect my plugins In-Reply-To: <1232443480.487.6.camel@mini.centricular.net> References: <21541362.post@talk.nabble.com> <3c1737210901191016s1848ee38q99b52ea8cec1bb2@mail.gmail.com> <21557477.post@talk.nabble.com> <1232443480.487.6.camel@mini.centricular.net> Message-ID: <21559965.post@talk.nabble.com> i am doing only gst_element_register(plugin, type_name, rank, type); i am not doing any other registration..... gst-inspect of mp4sp_dec is : [root at bghwf9861 testelement]# gst-inspect mp4sp_dec Factory Details: Long name: mp4sp Class: video/decoder Description: mp4sp Video Decoder Author(s): Rakesh Rank: unknown (260) Plugin Details: Name: rakesh_plugin Description: Rakesh Sharma Filename: /usr/local/lib/librakesh_plugin.so Version: 0.1 License: Proprietary Source module: rakesh_plugin Binary package: rakesh_plugin Origin URL: Rakesh GObject +----GstObject +----GstElement +----psf_mp4sp_dec Pad Templates: SINK template: 'sink' Availability: Always Capabilities: video/x-mp4-sp width: [ 0, 2147483647 ] height: [ 0, 2147483647 ] framerate: [ 1, 2147483647 ] SRC template: 'src' Availability: Always Capabilities: video/x-raw-yuv width: [ 1, 2147483647 ] height: [ 1, 2147483647 ] framerate: [ 0/1, 2147483647/1 ] Element Flags: no flags set Element Implementation: No loopfunc(), must be chain-based or not configured yet Has change_state() function: element_change_state_func Has custom save_thyself() function: element_save_thyself Has custom restore_thyself() function: element_restore_thyself Element has no clocking capabilities. Element has no indexing capabilities. Pads: SRC: 'src' Implementation: Has custom eventfunc(): 0xbbcf4c Pad Template: 'src' SINK: 'sink' Implementation: Has chainfunc(): 0xbbc269 Has custom eventfunc(): 0xbbcf73 Has bufferallocfunc(): video_dec_buffer_alloc Pad Template: 'sink' Element Properties: name : The name of the object flags: readable, writable String. Default: null Current: "mp4sp_dec0" -- View this message in context: http://www.nabble.com/How-to-register-Gstreamer-Plugin-with-Playbin-so-that-playbin-detect-my-plugins-tp21541362p21559965.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From bisht.sudarshan at gmail.com Tue Jan 20 11:22:30 2009 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Tue, 20 Jan 2009 15:52:30 +0530 Subject: [gst-devel] How to register Gstreamer Plugin with Playbin so that playbin detect my plugins In-Reply-To: <21559965.post@talk.nabble.com> References: <21541362.post@talk.nabble.com> <3c1737210901191016s1848ee38q99b52ea8cec1bb2@mail.gmail.com> <21557477.post@talk.nabble.com> <1232443480.487.6.camel@mini.centricular.net> <21559965.post@talk.nabble.com> Message-ID: <785339900901200222g61183847q4af355dc76c1a65c@mail.gmail.com> Why it is showing Rank: unknown (260) ? On Tue, Jan 20, 2009 at 3:20 PM, rakesh sharma wrote: > > i am doing only gst_element_register(plugin, type_name, rank, type); > i am not doing any other registration..... > > gst-inspect of mp4sp_dec is : > > [root at bghwf9861 testelement]# gst-inspect mp4sp_dec > Factory Details: > Long name: mp4sp > Class: video/decoder > Description: mp4sp Video Decoder > Author(s): Rakesh > Rank: unknown (260) > > Plugin Details: > Name: rakesh_plugin > Description: Rakesh Sharma > Filename: /usr/local/lib/librakesh_plugin.so > Version: 0.1 > License: Proprietary > Source module: rakesh_plugin > Binary package: rakesh_plugin > Origin URL: Rakesh > > GObject > +----GstObject > +----GstElement > +----psf_mp4sp_dec > > Pad Templates: > SINK template: 'sink' > Availability: Always > Capabilities: > video/x-mp4-sp > width: [ 0, 2147483647 ] > height: [ 0, 2147483647 ] > framerate: [ 1, 2147483647 ] > > SRC template: 'src' > Availability: Always > Capabilities: > video/x-raw-yuv > width: [ 1, 2147483647 ] > height: [ 1, 2147483647 ] > framerate: [ 0/1, 2147483647/1 ] > > > Element Flags: > no flags set > > Element Implementation: > No loopfunc(), must be chain-based or not configured yet > Has change_state() function: element_change_state_func > Has custom save_thyself() function: element_save_thyself > Has custom restore_thyself() function: element_restore_thyself > > Element has no clocking capabilities. > Element has no indexing capabilities. > > Pads: > SRC: 'src' > Implementation: > Has custom eventfunc(): 0xbbcf4c > Pad Template: 'src' > SINK: 'sink' > Implementation: > Has chainfunc(): 0xbbc269 > Has custom eventfunc(): 0xbbcf73 > Has bufferallocfunc(): video_dec_buffer_alloc > Pad Template: 'sink' > > Element Properties: > name : The name of the object > flags: readable, writable > String. Default: null Current: "mp4sp_dec0" > > -- > View this message in context: > http://www.nabble.com/How-to-register-Gstreamer-Plugin-with-Playbin-so-that-playbin-detect-my-plugins-tp21541362p21559965.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From t.i.m at zen.co.uk Tue Jan 20 11:23:06 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Tue, 20 Jan 2009 10:23:06 +0000 Subject: [gst-devel] How to register Gstreamer Plugin with Playbin so that playbin detect my plugins In-Reply-To: <21559965.post@talk.nabble.com> References: <21541362.post@talk.nabble.com> <3c1737210901191016s1848ee38q99b52ea8cec1bb2@mail.gmail.com> <21557477.post@talk.nabble.com> <1232443480.487.6.camel@mini.centricular.net> <21559965.post@talk.nabble.com> Message-ID: <1232446986.487.13.camel@mini.centricular.net> On Tue, 2009-01-20 at 01:50 -0800, rakesh sharma wrote: > [root at bghwf9861 testelement]# gst-inspect mp4sp_dec > Factory Details: > Long name: mp4sp > Class: video/decoder This should be: 'Codec/Decoder/Video' Cheers -Tim From thaytan at noraisin.net Tue Jan 20 12:17:09 2009 From: thaytan at noraisin.net (Jan Schmidt) Date: Tue, 20 Jan 2009 11:17:09 +0000 Subject: [gst-devel] rate change not very responsive In-Reply-To: <1232372905.6578.32.camel@art-laptop> References: <1232372905.6578.32.camel@art-laptop> Message-ID: <1232450229.1311.23.camel@fancy-ubuntu> On Mon, 2009-01-19 at 14:48 +0100, arturo castro wrote: > Hi > > I'm one of the developers of the openframeworks project: > http://openframeworks.cc , a framework for creative coding. Since now > we've been using some object oriented wrapper around FFMPEG as our video > backend in the linux version but it's being problematic and super slow > so i've decided to give gstreamer a try. > > With the new version of appsink in base everything has gone more or less > smooth except for one little problem. When i try to change a video > playing speed from a gui control like a slider or just mapping that > property to the mouse position, the video stops playing while speed is > changing and only continues playing again when speed stops changing. > > I'm using a pipeline like: > > filesrc location=... ! decodebin ! ffmpegcolorspace ! appsink > > also using signals in appsink to notify new buffers and polling the > pipeline for messages with: > > while(gst_bus_have_pending(bus)){ > GstMessage* msg = gst_bus_pop(bus); > ... > > as i cannot get callbacks from the bus since i cannot run in the > gstreamer loop because our framework already has it's own. > > I've tried with diferent seek flags but have more or less the same > result. > > This is the code i'm using to change speed directly mapped to gui > control: > > void ofVideoPlayer::setSpeed(speed) > GstFormat format = GST_FORMAT_TIME; > GstSeekFlags flags = (GstSeekFlags) (GST_SEEK_FLAG_FLUSH|GST_SEEK_FLAG_ACCURATE); > > if(!gst_element_seek(GST_ELEMENT(gstPipeline),speed, format, > flags, > GST_SEEK_TYPE_NONE, > 0, > GST_SEEK_TYPE_NONE, > -1)) { > ofLog(OF_WARNING,"GStreamer: unable to seek"); > } > } > > > also attached is the full code for our videoPlayer class. > > Is there anything i can do to make speed change more responsive, or is > this just a limitation of the current implementation of gstreamer? > How responsive it's going to be is a function of the pipeline mostly. You might want to check out the 'scrubby' example in gst-plugins-base/tests/examples/seek/scrubby.c - it does pretty much what you're looking for. There has been some discussion of how we might perform rate-changes-that-don't-change-the-playback-direction more quickly, but nothing has come of that yet. J. -- Jan Schmidt From thaytan at noraisin.net Tue Jan 20 13:24:44 2009 From: thaytan at noraisin.net (Jan Schmidt) Date: Tue, 20 Jan 2009 12:24:44 +0000 Subject: [gst-devel] Core/Base/Bad/Python released. Next step - moving to GIT. In-Reply-To: <1232421838.25255.6.camel@joder> References: <1232410808.1311.20.camel@fancy-ubuntu> <1232421838.25255.6.camel@joder> Message-ID: <1232454284.1311.24.camel@fancy-ubuntu> On Tue, 2009-01-20 at 04:23 +0100, Edward Hervey wrote: > Thanks Jan for the releases ! > > So for the missing info: > * I'll be grabbing an fdo admin to de-activate cvs (leaving only > anoncvs). No shortage of them here at LCA. > * Since this switch information has already been circulating for a few > weeks now, I propose to do that at 23:59 CET tonight (20th Jan). For > people in Europe it won't matter that way, but I'm currently in > Australia so leaves me a full day to make sure the migration goes > flawlessly before Europe wakes up again. +1 on the timing. J. > The rest is pretty straighforward, and the conversion/upload/activate > should take (with checks) around 4 hours. > > I don't think there's much more to say, > > See you the other side :) > > Edward > > On Tue, 2009-01-20 at 00:20 +0000, Jan Schmidt wrote: > > Now that the set of Core/Base/Python/Bad release are out of the way, the > > plan is to move to GIT over the next week. > > > > Edward will be managing the timing and activity of the actual migration > > to fit in with his week at LCA, but the general plan is: > > > > * Core/Base/Bad/Python stay frozen until further notice > > * Edward decides when he wants to cue up the migration and lets us know > > * At a designated time *all other* modules will freeze in CVS and CVS > > will be marked Read-only, so that there can be no commits while the > > migration takes place. > > * Everything moves to git in the standard locations on the > > freedesktop.org servers (which should be something like > > http://cgit.freedesktop.org/gstreamer/gstreamer, > > http://cgit.freedesktop.org/gstreamer/gst-plugins-base etc) > > * Everyone can start using git, and CVS is only useful as a read-only > > history. > > > > In case of unexpected disaster: > > All repositories are tagged with GIT_MIGRATION for the commit at which > > the conversions were done. If the worst comes to worst, we can then > > easily get all commits since that conversion as a series of unified-diff > > patches, make cvs read-write again, and commit them. > > > > Edward - over to you! > > > > Cheers, > > Jan. > -- Jan Schmidt From rahul.verma at sasken.com Tue Jan 20 14:43:09 2009 From: rahul.verma at sasken.com (Rahul Verma) Date: Tue, 20 Jan 2009 19:13:09 +0530 Subject: [gst-devel] 3gp demux plugin working on PC but not on target References: Message-ID: <89635EF2797FED4B9EF9769F80CEABC871F6AF@EXGZ01.sasken.com> Hi, >If you're in pull mode, you can (and should) pull from the upstream >element - which might, internally, use some sort of 'file handle', but >equally, it might not. Yes my plugin works in pull mode. But even in the reference plugin(ffmpeg_demux), I saw functions av_input_file_open, av_read_frame being used. I tried to go inside these functions and saw url_open and similar functions being used. Is it not as good as getting a file pointer directly and doing the file operations? What exactly do you mean when you say it must pull data? >gst_pad_push() will not generally return immediately. In this >pipeline, your demuxer is pushing buffers into a queue element. That >should be very fast usually (it just takes the appropriate locks, then >adds the buffer to the queue element's internal queue of buffers. >However, if the queue is full, it'll block until the queue stops being >full. >In your case, I don't know why you're seeing increasing time taken for >pushing on your pad - perhaps your timestamps or wrong, or your sink >or decoder isn't working correctly? I checked the timestamp values again and I compared with the timestamp values of the open source demuxer too. Since my 3gp reader gives time in milliseconds I multiply it with 1000000 to convert it into nanoseconds and give it to the sink. So the difference is very minor. Problem is still the same. _loop function of my parser plugin is taking too much time. Currently in my loop function I am doing the following steps: Seek to video offset. Allocate video buffer from video source pad Read from file into the buffer. Push the buffer. Same for audio. I don't understand why _push_pad function time goes to 100s of millisecond. Regards, Rahul Message: 3 Date: Mon, 19 Jan 2009 10:24:26 -0800 From: Michael Smith Subject: Re: [gst-devel] 3gp demux plugin working on PC but not on target To: Discussion of the development of GStreamer Message-ID: <3c1737210901191024i6232d6d2k89193d22189686bb at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 On Mon, Jan 19, 2009 at 5:59 AM, Rahul Verma wrote: > >>The whole concept is badly broken. Your element should not be poking >>at the implementation details of its peer - there's no reason to >>expect that the peer even HAS a "location" property. >> >>So, you should stop doing that entirely. I didn't check the actual >>code to see if there was an obvious bug in the actual implementation, >>because it really doesn't matter. > > Thanks Mike. Your point is valid. But for a 3gp reader, which works in a > pull mode, the file operations has to be done by the reader, since it > knows the parsing logic. If my understanding is correct, then it has to get > the file handle. What is the alternative way to get the file handle then? If you're in pull mode, you can (and should) pull from the upstream element - which might, internally, use some sort of 'file handle', but equally, it might not. To do this, you use gst_pad_pull_range() on your element's sinkpad. You MUST NOT directly use any sort of file handle. > > my pipeline is > > filesrc ! 3gpdemux name=demux demux.! queue ! videodecoder ! > videosink demux.! queue ! audiodecoder ! osssink > > Is gst_pad_push a synchronous call and will be blocking? Initially all is > well, but as soon as > rendering starts, gst_push_pad time starts increasing. gst_pad_push() will not generally return immediately. In this pipeline, your demuxer is pushing buffers into a queue element. That should be very fast usually (it just takes the appropriate locks, then adds the buffer to the queue element's internal queue of buffers. However, if the queue is full, it'll block until the queue stops being full. In your case, I don't know why you're seeing increasing time taken for pushing on your pad - perhaps your timestamps or wrong, or your sink or decoder isn't working correctly? Mike SASKEN BUSINESS DISCLAIMER ------------------------- This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, Disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email From tim.hilden at googlemail.com Tue Jan 20 17:09:32 2009 From: tim.hilden at googlemail.com (Tim Hilden) Date: Tue, 20 Jan 2009 17:09:32 +0100 Subject: [gst-devel] element specific bus messages Message-ID: <1232467772.6039.40.camel@calculus> hi there, playing around with the (sparsly documented) Gstreamer a couple of questions came to my mind. This one is about the element specific bus messages. Getting a message of type GST_MESSAGE_ELEMENT I wonder where I can find the "specific element's documentation" that is mentioned in the API doc. The online doc of elements like xvimagesink or jpegenc don't contain any information on what the element's bus messages contain. So where can I find that information? Tim From rbultje at ronald.bitfreak.net Tue Jan 20 18:25:06 2009 From: rbultje at ronald.bitfreak.net (Ronald S. Bultje) Date: Tue, 20 Jan 2009 12:25:06 -0500 Subject: [gst-devel] Fwd: GStreamer subtitles task In-Reply-To: <537e39c90901200906t2d351e31ndc9f73a7659a17cc@mail.gmail.com> References: <537e39c90901200906t2d351e31ndc9f73a7659a17cc@mail.gmail.com> Message-ID: <34539a480901200925l56ed1fdexd1ee34a84808679e@mail.gmail.com> Hi guys, can you please answer this guy's question? Also, the tasks page quite clearly needs some updating... Ronald ---------- Forwarded message ---------- From: Jordi Jaen Pallares Date: Tue, Jan 20, 2009 at 12:06 PM Subject: GStreamer subtitles task To: thomas at apestaart.org, rbultje at ronald.bitfreak.net Hi ! do you have someone to help you with the subtitles issue in GStreamer ? I am working for an european project (http://www.ict-ccast.eu/) that will need subtitle support. Since I am also pushing GStreamer as a principal streaming engine in the project, I would like to contribute to the subtitles task as announced in http://gstreamer.freedesktop.org/tasks/gstreamer.html Please let me know if u still need help on that and what is required. I would be interested in some kind of plug-in to blend some text as subtitles on a video. Best regards, Jordi PS. some time ago I sent a small patch for the GStreamer MPEG-TS RTP packetizer plugin From arturo at openframeworks.cc Tue Jan 20 18:32:26 2009 From: arturo at openframeworks.cc (arturo castro) Date: Tue, 20 Jan 2009 18:32:26 +0100 Subject: [gst-devel] rate change not very responsive In-Reply-To: <1232450229.1311.23.camel@fancy-ubuntu> References: <1232372905.6578.32.camel@art-laptop> <1232450229.1311.23.camel@fancy-ubuntu> Message-ID: <1232472746.23113.46.camel@art-laptop> Hi Jan thanks very much for your answer. the thing is i've been trying the scrubby example and at least for me it's super unresponsive. when trying to seek or change speed the app freeze for some seconds. also tried that code in my app and the result is more or less the same. in my code I've managed to have a very responsive speed change (even with negative speeds) by always seeking to the current position on speed changes, don't know if this is very correct in gstreamer terms though :) something like: GstFormat format = GST_FORMAT_TIME; GstSeekFlags flags = (GstSeekFlags) (GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE); gint64 pos; gst_element_set_state (gstPipeline, GST_STATE_PLAYING); if(speed>0){ gst_element_seek(GST_ELEMENT(gstPipeline),speed, format, flags, GST_SEEK_TYPE_SET, pos, GST_SEEK_TYPE_SET, -1)) }else{ gst_element_seek(GST_ELEMENT(gstPipeline),speed, format, flags, GST_SEEK_TYPE_SET, 0, GST_SEEK_TYPE_SET, pos)) } i've also tried this code with the scrubby example, with same results, no freezes and almost immediate response to the slider even for negative speeds. I've attached a modified scrubby.c thanks El mar, 20-01-2009 a las 11:17 +0000, Jan Schmidt escribi?: > On Mon, 2009-01-19 at 14:48 +0100, arturo castro wrote: > > Hi > > > > I'm one of the developers of the openframeworks project: > > http://openframeworks.cc , a framework for creative coding. Since now > > we've been using some object oriented wrapper around FFMPEG as our video > > backend in the linux version but it's being problematic and super slow > > so i've decided to give gstreamer a try. > > > > With the new version of appsink in base everything has gone more or less > > smooth except for one little problem. When i try to change a video > > playing speed from a gui control like a slider or just mapping that > > property to the mouse position, the video stops playing while speed is > > changing and only continues playing again when speed stops changing. > > > > I'm using a pipeline like: > > > > filesrc location=... ! decodebin ! ffmpegcolorspace ! appsink > > > > also using signals in appsink to notify new buffers and polling the > > pipeline for messages with: > > > > while(gst_bus_have_pending(bus)){ > > GstMessage* msg = gst_bus_pop(bus); > > ... > > > > as i cannot get callbacks from the bus since i cannot run in the > > gstreamer loop because our framework already has it's own. > > > > I've tried with diferent seek flags but have more or less the same > > result. > > > > This is the code i'm using to change speed directly mapped to gui > > control: > > > > void ofVideoPlayer::setSpeed(speed) > > GstFormat format = GST_FORMAT_TIME; > > GstSeekFlags flags = (GstSeekFlags) (GST_SEEK_FLAG_FLUSH|GST_SEEK_FLAG_ACCURATE); > > > > if(!gst_element_seek(GST_ELEMENT(gstPipeline),speed, format, > > flags, > > GST_SEEK_TYPE_NONE, > > 0, > > GST_SEEK_TYPE_NONE, > > -1)) { > > ofLog(OF_WARNING,"GStreamer: unable to seek"); > > } > > } > > > > > > also attached is the full code for our videoPlayer class. > > > > Is there anything i can do to make speed change more responsive, or is > > this just a limitation of the current implementation of gstreamer? > > > > How responsive it's going to be is a function of the pipeline mostly. > You might want to check out the 'scrubby' example in > gst-plugins-base/tests/examples/seek/scrubby.c - it does pretty much > what you're looking for. > > There has been some discussion of how we might perform > rate-changes-that-don't-change-the-playback-direction more quickly, but > nothing has come of that yet. > > J. > -------------- next part -------------- A non-text attachment was scrubbed... Name: scrubby.c Type: text/x-csrc Size: 14722 bytes Desc: not available URL: From msmith at xiph.org Tue Jan 20 19:02:00 2009 From: msmith at xiph.org (Michael Smith) Date: Tue, 20 Jan 2009 10:02:00 -0800 Subject: [gst-devel] add a plugin In-Reply-To: <49756AA0.8090909@nestgroup.net> References: <49756471.60105@nestgroup.net> <785339900901192158o7ffe4087vd2e54e8177c90661@mail.gmail.com> <49756AA0.8090909@nestgroup.net> Message-ID: <3c1737210901201002t275fbb43x1a6c00c2edd12164@mail.gmail.com> > On Tue, Jan 20, 2009 at 11:13 AM, vaisakh.n wrote: >> >> Hi, >> I have a decoder plugin's library file ( .so file ) >> how can i add it to gstreamer plugin registry.. >> please help me >> what are the steps i need to follow? >> advance thanks > > Hi, > no its not working > plugin is written by us .its does not have any api to link with > gstreamer.. Well, if your shared library isn't a gstreamer plugin, then obviously gstreamer can't load it. You need to write a gstreamer plugin. You should read the GStreamer Plugin Writers Guide to learn how to do this. Mike From msmith at xiph.org Tue Jan 20 19:09:18 2009 From: msmith at xiph.org (Michael Smith) Date: Tue, 20 Jan 2009 10:09:18 -0800 Subject: [gst-devel] How to register Gstreamer Plugin with Playbin so that playbin detect my plugins In-Reply-To: <785339900901200222g61183847q4af355dc76c1a65c@mail.gmail.com> References: <21541362.post@talk.nabble.com> <3c1737210901191016s1848ee38q99b52ea8cec1bb2@mail.gmail.com> <21557477.post@talk.nabble.com> <1232443480.487.6.camel@mini.centricular.net> <21559965.post@talk.nabble.com> <785339900901200222g61183847q4af355dc76c1a65c@mail.gmail.com> Message-ID: <3c1737210901201009s219c7e52rac3a81b8056affbd@mail.gmail.com> On Tue, Jan 20, 2009 at 2:22 AM, sudarshan bisht wrote: > Why it is showing Rank: unknown (260) ? There are a few "named" ranks - minimal, normal, maximum, and so on. But rank is just a number, so you can have intermediate values too - which this plugin has. That's actually fine. > > On Tue, Jan 20, 2009 at 3:20 PM, rakesh sharma > wrote: >> >> i am doing only gst_element_register(plugin, type_name, rank, type); >> i am not doing any other registration..... >> >> gst-inspect of mp4sp_dec is : >> >> [root at bghwf9861 testelement]# gst-inspect mp4sp_dec >> Factory Details: >> Long name: mp4sp >> Class: video/decoder >> Description: mp4sp Video Decoder >> Author(s): Rakesh >> Rank: unknown (260) >> >> Plugin Details: >> Name: rakesh_plugin >> Description: Rakesh Sharma >> Filename: /usr/local/lib/librakesh_plugin.so >> Version: 0.1 >> License: Proprietary >> Source module: rakesh_plugin >> Binary package: rakesh_plugin >> Origin URL: Rakesh >> >> GObject >> +----GstObject >> +----GstElement >> +----psf_mp4sp_dec >> >> Pad Templates: >> SINK template: 'sink' >> Availability: Always >> Capabilities: >> video/x-mp4-sp >> width: [ 0, 2147483647 ] >> height: [ 0, 2147483647 ] >> framerate: [ 1, 2147483647 ] This looks wrong in several ways: "video/x-mp4-sp" isn't, as far as I know, used anywhere in gstreamer. Of course, you might be using a custom demuxer that produces it, but you didn't mention anything like that. So I assume your decoder isn't actually compatible with any types that any demuxers might output. Additionally, framerate in gstreamer is a fraction, you have it here as an integer. Make sure you fix up the other things mentioned by other people too. Mike From msmith at xiph.org Tue Jan 20 19:13:45 2009 From: msmith at xiph.org (Michael Smith) Date: Tue, 20 Jan 2009 10:13:45 -0800 Subject: [gst-devel] 3gp demux plugin working on PC but not on target In-Reply-To: <89635EF2797FED4B9EF9769F80CEABC871F6AF@EXGZ01.sasken.com> References: <89635EF2797FED4B9EF9769F80CEABC871F6AF@EXGZ01.sasken.com> Message-ID: <3c1737210901201013v6529e797pa422292ff33a0435@mail.gmail.com> On Tue, Jan 20, 2009 at 5:43 AM, Rahul Verma wrote: > Hi, > > >>If you're in pull mode, you can (and should) pull from the upstream >>element - which might, internally, use some sort of 'file handle', but >>equally, it might not. > > Yes my plugin works in pull mode. But even in the reference > plugin(ffmpeg_demux), I saw functions av_input_file_open, av_read_frame > being used. I tried to go inside these functions and saw url_open and > similar functions being used. Is it not as good as getting a file > pointer directly and doing the file operations? What exactly do you mean > when you say it must pull data? ffmpeg_demux should not be used as a reference plugin for anything. It's not a good example to use at all. Don't use it for any comparisons; the ffmpeg demuxers are known to work poorly (and in many cases not work at all). I don't actually know what it does internally - I haven't looked - but a gstreamer demuxer plugin can't act on a file directly, because it doesn't even know if there IS a file involved at all. By "it must pull data" I mean that to get data from the upstream element (in pull mode), you must use gst_pad_pull_range(). > >>gst_pad_push() will not generally return immediately. In this >>pipeline, your demuxer is pushing buffers into a queue element. That >>should be very fast usually (it just takes the appropriate locks, then >>adds the buffer to the queue element's internal queue of buffers. >>However, if the queue is full, it'll block until the queue stops being >>full. > >>In your case, I don't know why you're seeing increasing time taken for >>pushing on your pad - perhaps your timestamps or wrong, or your sink >>or decoder isn't working correctly? > > I checked the timestamp values again and I compared with the timestamp > values of the open source demuxer too. Since my 3gp reader gives time in > milliseconds I multiply it with 1000000 to convert it into nanoseconds > and give it to the sink. So the difference is very minor. Problem is > still the same. _loop function of my parser plugin is taking too much > time. > > Currently in my loop function I am doing the following steps: > > Seek to video offset. > Allocate video buffer from video source pad > Read from file into the buffer. > Push the buffer. > > Same for audio. > > I don't understand why _push_pad function time goes to 100s of > millisecond. I don't know either. Sorry, I can only give general suggestions, since you're not writing an open source plugin. Looking at the debug output (GST_DEBUG) might help you. Mike From msmith at xiph.org Tue Jan 20 19:16:11 2009 From: msmith at xiph.org (Michael Smith) Date: Tue, 20 Jan 2009 10:16:11 -0800 Subject: [gst-devel] element specific bus messages In-Reply-To: <1232467772.6039.40.camel@calculus> References: <1232467772.6039.40.camel@calculus> Message-ID: <3c1737210901201016p35764100v985520ff16b47d3d@mail.gmail.com> On Tue, Jan 20, 2009 at 8:09 AM, Tim Hilden wrote: > hi there, > > playing around with the (sparsly documented) Gstreamer a couple of > questions came to my mind. > > This one is about the element specific bus messages. Getting a message > of type GST_MESSAGE_ELEMENT I wonder where I can find the "specific > element's documentation" that is mentioned in the API doc. The online > doc of elements like xvimagesink or jpegenc don't contain any > information on what the element's bus messages contain. So where can I > find that information? That would mean a) those elements don't have any element-specific messages, or b) they're missing documentation for any they might send. In this case, they just don't have any element-specific messages (though xvimagesink will send a "prepare-xwindow-id" message - as documented in the GstXOverlay interface, which it implements. Mike From wim.taymans at gmail.com Tue Jan 20 19:22:11 2009 From: wim.taymans at gmail.com (Wim Taymans) Date: Tue, 20 Jan 2009 19:22:11 +0100 Subject: [gst-devel] element specific bus messages In-Reply-To: <1232467772.6039.40.camel@calculus> References: <1232467772.6039.40.camel@calculus> Message-ID: those elements don't implement element specific messages. Take a look at the spectrum element here for an element that implements element messages: http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-spectrum.html Wim On Tue, Jan 20, 2009 at 5:09 PM, Tim Hilden wrote: > hi there, > > playing around with the (sparsly documented) Gstreamer a couple of > questions came to my mind. > > This one is about the element specific bus messages. Getting a message > of type GST_MESSAGE_ELEMENT I wonder where I can find the "specific > element's documentation" that is mentioned in the API doc. The online > doc of elements like xvimagesink or jpegenc don't contain any > information on what the element's bus messages contain. So where can I > find that information? > > Tim > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From maximlevitsky at gmail.com Tue Jan 20 23:29:30 2009 From: maximlevitsky at gmail.com (Maxim Levitsky) Date: Wed, 21 Jan 2009 00:29:30 +0200 Subject: [gst-devel] Few questions about status of gstreamer Message-ID: <1232490570.15685.35.camel@maxim-laptop> Hi, I was tinkering with gstreamer lately, and I find it qute nice platform. I would like to know where current development is headed, and what to expect. While the concept of gstreamer is great, the current implementation isn't that great, I would like to point at following things: - web live streams playback is very bad compared even to other linux media players, as a test case visit any web site that offers list of live web channels, and you will be able to watch maybe 8% og them or so, this is very sad. I know that there are property features, but many streams are yet playable in vlc or mplayer. I wish there were a general database of links to such streams that will be visited by gstreamer/totem developers as this will encourage them to fix support for that. Also such database will ease the regression testing. ( for example ms switched to asf over rtsp, and I see some streams that use it, but gstreamer will just fail as it will think it is a mms stream also support for wmv over rtsp isn't in gstreamer, patches are aviable, but it doesn't seem that they will be mereged soon. sad. I have no free time yet ( might have some in near future ) to do that myself. ( note that I am speaking about web sites that offer list of live streams, as they provide direct links without fancy media player testing, this rules out web page incompatibilities due to sites using windows only player controls) - gstreamer elements are far from beeng generic: look at that for example : maxim at maxim-laptop:~$ gst-inspect xvidenc | grep video Long name: XviD video encoder video/x-raw-yuv video/x-raw-rgb video/x-raw-rgb video/x-raw-rgb video/x-raw-rgb video/x-raw-rgb video/x-raw-rgb video/x-raw-rgb video/x-xvid as you see it supports only 'video/x-xvid' on output, but why, isn't xvid output plain mpeg-4 ? on the other hand maxim at maxim-laptop:~$ gst-inspect ffmux_mp4 | grep video video/quicktime SINK template: 'video_%d' video/mpeg video/x-divx video/x-h264 this is just one of may examples. in other words currently muxers are very picky about formats they can handle although underlying formats can hold much more codecs AVI for example can pretty much hold any codec that have fourcc code. also note that sinks of muxers sometimes are named video_[0-9] and sometimes video_[0-9][0-9] while it is possible to avoid referencing those names directly, still it would be better to have standard naming - There are many pipeline stalls. it might be no joy to sit for a 4 hours to figure out 'where did I forgot to put that 'queue' also without magic 'videorate' that drops frames on demand (right?) it is very hard/impossible to create working pipeline, (not to mention that you absolutely need queues after each encoder) also generally there is no way to copy a raw stream from one container to another. I for example tried to put asf stream (captured into file) ( gstream doesn't seem to support seeking in it) to avi, which is possible and is done nicely by mencoder. Gstreamer just stalls. I figured out that the problem was that intially input stream contained one big (~ 27 KB) chunk of video data, and then audio, but this made asfdemux stall, as it can't pull both video and audio data seperatly In other words demuxers can easily stall if input stream isn't evenly interleaved. Best regards, Maxim Levitsky PS: do you know whenever gstreamer can add an index to avi file? I use this command to record a live asf stream: > #source description > export SOURCE_PROTOCOL=mmssrc > export SOURCE_DEMUXER=asfdemux > export SOURCE_AUDIO_CODEC=ffdec_wmav2 > export SOURCE_VIDEO_CODEC=ffdec_wmv3 > > #destanation description > export DEST_PROTOCOL=filesink > export DEST_MUXER=avimux > export DEST_AUDIO_CODEC=lame > export DEST_VIDEO_CODEC=xvidenc > gst-launch \ > $SOURCE_PROTOCOL location=$SOURCE_LOCATION ! queue ! $SOURCE_DEMUXER name=demuxer \ > \ > demuxer. ! queue ! $SOURCE_VIDEO_CODEC ! videorate ! postproc_default ! tee name=raw_video \ > demuxer. ! queue ! $SOURCE_AUDIO_CODEC ! tee name=raw_audio \ > \ > raw_video. ! queue leaky=1 ! $DEST_VIDEO_CODEC ! queue ! $DEST_MUXER name=muxer \ > raw_audio. ! queue leaky=1 ! $DEST_AUDIO_CODEC ! queue ! muxer. \ > \ > raw_video. ! queue ! gconfvideosink \ > raw_audio. ! queue ! gconfaudiosink \ > \ > muxer. ! $DEST_PROTOCOL location=$DEST_LOCATION \ It works, but resulting file can be seeking to non key-frames, which obliviously make seeking very unpleasant. (since muxers are limited to such small number of codecs, can't they know which frames are key and which not ?) From bilboed at gmail.com Wed Jan 21 02:21:33 2009 From: bilboed at gmail.com (Edward Hervey) Date: Wed, 21 Jan 2009 02:21:33 +0100 Subject: [gst-devel] Core/Base/Bad/Python released. Next step - moving to GIT. In-Reply-To: <1232454284.1311.24.camel@fancy-ubuntu> References: <1232410808.1311.20.camel@fancy-ubuntu> <1232421838.25255.6.camel@joder> <1232454284.1311.24.camel@fancy-ubuntu> Message-ID: <1232500893.7969.2.camel@joder> Update: * CVS is now read-only (anoncvs) * conversion is currently underway I expect to be able to push the converted repos in 2 hours Edward On Tue, 2009-01-20 at 12:24 +0000, Jan Schmidt wrote: > On Tue, 2009-01-20 at 04:23 +0100, Edward Hervey wrote: > > Thanks Jan for the releases ! > > > > So for the missing info: > > * I'll be grabbing an fdo admin to de-activate cvs (leaving only > > anoncvs). No shortage of them here at LCA. > > * Since this switch information has already been circulating for a few > > weeks now, I propose to do that at 23:59 CET tonight (20th Jan). For > > people in Europe it won't matter that way, but I'm currently in > > Australia so leaves me a full day to make sure the migration goes > > flawlessly before Europe wakes up again. > > +1 on the timing. > > J. > > > The rest is pretty straighforward, and the conversion/upload/activate > > should take (with checks) around 4 hours. > > > > I don't think there's much more to say, > > > > See you the other side :) > > > > Edward > > > > On Tue, 2009-01-20 at 00:20 +0000, Jan Schmidt wrote: > > > Now that the set of Core/Base/Python/Bad release are out of the way, the > > > plan is to move to GIT over the next week. > > > > > > Edward will be managing the timing and activity of the actual migration > > > to fit in with his week at LCA, but the general plan is: > > > > > > * Core/Base/Bad/Python stay frozen until further notice > > > * Edward decides when he wants to cue up the migration and lets us know > > > * At a designated time *all other* modules will freeze in CVS and CVS > > > will be marked Read-only, so that there can be no commits while the > > > migration takes place. > > > * Everything moves to git in the standard locations on the > > > freedesktop.org servers (which should be something like > > > http://cgit.freedesktop.org/gstreamer/gstreamer, > > > http://cgit.freedesktop.org/gstreamer/gst-plugins-base etc) > > > * Everyone can start using git, and CVS is only useful as a read-only > > > history. > > > > > > In case of unexpected disaster: > > > All repositories are tagged with GIT_MIGRATION for the commit at which > > > the conversions were done. If the worst comes to worst, we can then > > > easily get all commits since that conversion as a series of unified-diff > > > patches, make cvs read-write again, and commit them. > > > > > > Edward - over to you! > > > > > > Cheers, > > > Jan. > > From bilboed at gmail.com Wed Jan 21 07:15:25 2009 From: bilboed at gmail.com (Edward Hervey) Date: Wed, 21 Jan 2009 07:15:25 +0100 Subject: [gst-devel] GIT conversion ... DONE Message-ID: <1232518526.7969.12.camel@joder> Hello hackers, All the main GStreamer modules are now converted to git and up. You can view them here http://cgit.freedesktop.org/gstreamer/ The url to access each module is at the bottom of each of those modules. The next steps are: * Finishing/activating the post-commit hook to send mails to the commit mailing-list * Convert the GStreamer website pages containing cvs information to mention the new git urls/usages. * Update the GitMigration wiki page, and migrate some information from that page to gstreamer/docs/ * Hacking I have currently got to leave for a conference dinner at LCA, but will be back online after that to finish those. The last step (hacking) can be done without me being online :) Just remember that the decision was made that we no longer commit changes to the ChangeLog file but only in the commit entry. The ChangeLog files will be automatically generated at (pre-)release time using a script mentionned in the GitMigration wiki page. Thanks to Daniel Stone from freedesktop.org for the help in the administration side of things. CVS is dead, long live GIT ! Edward From rakesh2.sharma at aricent.com Wed Jan 21 08:00:05 2009 From: rakesh2.sharma at aricent.com (rakesh sharma) Date: Tue, 20 Jan 2009 23:00:05 -0800 (PST) Subject: [gst-devel] How to register Gstreamer Plugin with Playbin so that playbin detect my plugins In-Reply-To: <3c1737210901201009s219c7e52rac3a81b8056affbd@mail.gmail.com> References: <21541362.post@talk.nabble.com> <3c1737210901191016s1848ee38q99b52ea8cec1bb2@mail.gmail.com> <21557477.post@talk.nabble.com> <1232443480.487.6.camel@mini.centricular.net> <21559965.post@talk.nabble.com> <785339900901200222g61183847q4af355dc76c1a65c@mail.gmail.com> <3c1737210901201009s219c7e52rac3a81b8056affbd@mail.gmail.com> Message-ID: <21578039.post@talk.nabble.com> i am using the caps like this src caps src_template = gst_pad_template_new("src", GST_PAD_SRC, GST_PAD_ALWAYS, gst_caps_from_string("audio/x-raw-int, " "rate = (int) [ 1, MAX ], " "endianness = (int) BYTE_ORDER, " "signed = (boolean) { true, false }, " "channels = (int) { 1, 2}, " "width = (int) { 8, 16 }, " "depth = (int) { 8, 16 }")); and sink caps sink_template = gst_pad_template_new("sink", GST_PAD_SINK, GST_PAD_ALWAYS, klass->dec->sink_caps); dec->sink_caps = gst_caps_from_string("audio/x-mp3-unknown, " "outsize = (int) [ 1, MAX ], " "layer = (int) {1,2,3}, " "version = (int) {1,2,25}, " "rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }, " "channels = (int) { 1, 2 }"); If i am doing it GST_STATIC_CAPS ("ANY")); then its working.... might be the caps are not proper .... can u suggest me the proper caps for mp3 parser and the proper way...!! -- View this message in context: http://www.nabble.com/How-to-register-Gstreamer-Plugin-with-Playbin-so-that-playbin-detect-my-plugins-tp21541362p21578039.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From rakesh2.sharma at aricent.com Wed Jan 21 08:01:13 2009 From: rakesh2.sharma at aricent.com (rakesh sharma) Date: Tue, 20 Jan 2009 23:01:13 -0800 (PST) Subject: [gst-devel] How to register Gstreamer Plugin with Playbin so that playbin detect my plugins Message-ID: <21578039.post@talk.nabble.com> i am using the caps like this src caps src_template = gst_pad_template_new("src", GST_PAD_SRC, GST_PAD_ALWAYS, gst_caps_from_string("audio/x-raw-int, " "rate = (int) [ 1, MAX ], " "endianness = (int) BYTE_ORDER, " "signed = (boolean) { true, false }, " "channels = (int) { 1, 2}, " "width = (int) { 8, 16 }, " "depth = (int) { 8, 16 }")); and sink caps klass->dec->sink_caps = gst_caps_from_string("audio/x-mp3-unknown, " "outsize = (int) [ 1, MAX ], " "layer = (int) {1,2,3}, " "version = (int) {1,2,25}, " "rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }, " "channels = (int) { 1, 2 }"); sink_template = gst_pad_template_new("sink", GST_PAD_SINK, GST_PAD_ALWAYS, klass->dec->sink_caps); If i am doing it GST_STATIC_CAPS ("ANY")); then its working.... might be the caps are not proper .... can u suggest me the proper caps for mp3 parser and the proper way...!! -- View this message in context: http://www.nabble.com/How-to-register-Gstreamer-Plugin-with-Playbin-so-that-playbin-detect-my-plugins-tp21541362p21578039.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From mediadevel at gmail.com Wed Jan 21 08:29:21 2009 From: mediadevel at gmail.com (john david) Date: Wed, 21 Jan 2009 12:59:21 +0530 Subject: [gst-devel] how to capture and play with out any jerks in ts/avi Message-ID: Hi, When i captured and encoding to mpegts using Gstreamer plug in , v4l2src, After encode and played that file there is jerks in video. When i captured and encoded only video elementary stream there is no jerks. Is there any modifications that i have to do? if ,then what modifications to do. In gstv4l2src.c i changed framerate to 7.5fps to 25fps gst_structure_fixate_field_nearest_fraction( structure, "framerate", 15, 2), to gst_structure_fixate_field_nearest_fraction( structure, "framerate", 25, 1). Still the jerks are coming. So please give me any idea about this. Regards, John -------------- next part -------------- An HTML attachment was scrubbed... URL: From t.i.m at zen.co.uk Wed Jan 21 09:55:28 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Wed, 21 Jan 2009 08:55:28 +0000 Subject: [gst-devel] How to register Gstreamer Plugin with Playbin so that playbin detect my plugins In-Reply-To: <21578039.post@talk.nabble.com> References: <21578039.post@talk.nabble.com> Message-ID: <1232528128.13038.4.camel@mini.centricular.net> On Tue, 2009-01-20 at 23:01 -0800, rakesh sharma wrote: > and sink caps > klass->dec->sink_caps = gst_caps_from_string("audio/x-mp3-unknown, " > "outsize = (int) [ 1, MAX ], " > "layer = (int) {1,2,3}, " > "version = (int) {1,2,25}, " > "rate = (int) { 8000, 11025, > 12000, 16000, 22050, 24000, 32000, 44100, 48000 }, " > "channels = (int) { 1, 2 }"); > sink_template = gst_pad_template_new("sink", GST_PAD_SINK, GST_PAD_ALWAYS, > klass->dec->sink_caps); > > If i am doing it GST_STATIC_CAPS ("ANY")); > then its working.... might be the caps are not proper .... > can u suggest me the proper caps for mp3 parser and the proper way...!! Where did you get the audio/x-mp3-unknown caps from? If you want to be compatible with other parts of GStreamer, like the typefinders, you can't just make up new media types (unless you also register some corresponding typefinder of your own as well). Have a look at one of the existing mp3 decoders/parsers (mad, mp3parse/mpegaudioparse) for the right caps. Cheers -Tim From weian.chen at intel.com Wed Jan 21 10:06:33 2009 From: weian.chen at intel.com (Chen, Weian) Date: Wed, 21 Jan 2009 17:06:33 +0800 Subject: [gst-devel] Does Gstreamer has a element that can split one stream into two Message-ID: Hi all: If I have a RGB format stream (decoded from H.264 through H.264 decoder element as an example) and I want display (link to xvimagesink) and save on disk (filesink) at one time, does Gstreamer has such filter element which is used to link above H.264 decoder element and xvimagesink/ filesink? Thanks in advanced. Weian -------------- next part -------------- An HTML attachment was scrubbed... URL: From E3423C at motorola.com Wed Jan 21 10:10:23 2009 From: E3423C at motorola.com (Zhao Liang-E3423C) Date: Wed, 21 Jan 2009 17:10:23 +0800 Subject: [gst-devel] Does Gstreamer has a element that can split one stream into two In-Reply-To: References: Message-ID: you can try tee. Zhao Liang ________________________________ From: Chen, Weian [mailto:weian.chen at intel.com] Sent: Wednesday, January 21, 2009 5:07 PM To: gstreamer-devel at lists.sourceforge.net Subject: [gst-devel] Does Gstreamer has a element that can split one stream into two Hi all: If I have a RGB format stream (decoded from H.264 through H.264 decoder element as an example) and I want display (link to xvimagesink) and save on disk (filesink) at one time, does Gstreamer has such filter element which is used to link above H.264 decoder element and xvimagesink/ filesink? Thanks in advanced. Weian -------------- next part -------------- An HTML attachment was scrubbed... URL: From t.i.m at zen.co.uk Wed Jan 21 10:13:09 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Wed, 21 Jan 2009 09:13:09 +0000 Subject: [gst-devel] GIT conversion ... DONE In-Reply-To: <1232518526.7969.12.camel@joder> References: <1232518526.7969.12.camel@joder> Message-ID: <1232529189.13038.13.camel@mini.centricular.net> On Wed, 2009-01-21 at 07:15 +0100, Edward Hervey wrote: > CVS is dead, long live GIT ! Yay! \o/ For those who still have modifications in their CVS developer checkouts, you can convert them to anoncvs checkouts with something like this: find -type d -name CVS -exec sed -i -e 's/:ext:.*@cvs.freedesktop.org/:pserver:anoncvs at anoncvs.freedesktop.org/' {}/Root \; touch ~/.cvspass Cheers -Tim From t.i.m at zen.co.uk Wed Jan 21 10:23:29 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Wed, 21 Jan 2009 09:23:29 +0000 Subject: [gst-devel] Does Gstreamer has a element that can split one stream into two In-Reply-To: References: Message-ID: <1232529809.13038.18.camel@mini.centricular.net> On Wed, 2009-01-21 at 17:06 +0800, Chen, Weian wrote: > If I have a RGB format stream (decoded from H.264 through H.264 > decoder element as an example) and I want display (link to > xvimagesink) and save on disk (filesink) at one time, does Gstreamer > has such filter element which is used to link above H.264 decoder > element and xvimagesink/ filesink? The 'tee' element might do what you want. Don't forget to add a queue after tee for each stream, e.g.: gst-launch-0.10 videotestsrc ! tee name=t \ t. ! queue ! ximagesink \ t. ! queue ! ximagesink Cheers -Tim From thaytan at noraisin.net Wed Jan 21 10:49:24 2009 From: thaytan at noraisin.net (Jan Schmidt) Date: Wed, 21 Jan 2009 09:49:24 +0000 Subject: [gst-devel] GIT conversion ... DONE In-Reply-To: <1232518526.7969.12.camel@joder> References: <1232518526.7969.12.camel@joder> Message-ID: <1232531364.4111.1.camel@fancy-ubuntu> On Wed, 2009-01-21 at 07:15 +0100, Edward Hervey wrote: > Hello hackers, > > All the main GStreamer modules are now converted to git and up. > > You can view them here http://cgit.freedesktop.org/gstreamer/ > The url to access each module is at the bottom of each of those > modules. > Hey hey, Thanks for getting the conversion done and everything online. Jonathon pointed to this commit as problematic: http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=a3e95ebfc9586534fa26d7ea59f34987f6e78ad1 The author attribution is incorrect. I guess where there's one, there's others too. How would we go about fixing that now? Is it worth fixing? J. > The next steps are: > * Finishing/activating the post-commit hook to send mails to the > commit mailing-list > * Convert the GStreamer website pages containing cvs information to > mention the new git urls/usages. > * Update the GitMigration wiki page, and migrate some information from > that page to gstreamer/docs/ > * Hacking > > I have currently got to leave for a conference dinner at LCA, but will > be back online after that to finish those. > > The last step (hacking) can be done without me being online :) > Just remember that the decision was made that we no longer commit > changes to the ChangeLog file but only in the commit entry. The > ChangeLog files will be automatically generated at (pre-)release time > using a script mentionned in the GitMigration wiki page. > > Thanks to Daniel Stone from freedesktop.org for the help in the > administration side of things. > > CVS is dead, long live GIT ! > > Edward > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Jan Schmidt From felipe.contreras at gmail.com Wed Jan 21 11:00:38 2009 From: felipe.contreras at gmail.com (Felipe Contreras) Date: Wed, 21 Jan 2009 12:00:38 +0200 Subject: [gst-devel] GIT conversion ... DONE In-Reply-To: <1232531364.4111.1.camel@fancy-ubuntu> References: <1232518526.7969.12.camel@joder> <1232531364.4111.1.camel@fancy-ubuntu> Message-ID: <94a0d4530901210200q7df695a6y2001a661942bb8de@mail.gmail.com> On Wed, Jan 21, 2009 at 11:49 AM, Jan Schmidt wrote: > On Wed, 2009-01-21 at 07:15 +0100, Edward Hervey wrote: >> Hello hackers, >> >> All the main GStreamer modules are now converted to git and up. >> >> You can view them here http://cgit.freedesktop.org/gstreamer/ >> The url to access each module is at the bottom of each of those >> modules. >> > > Hey hey, > > Thanks for getting the conversion done and everything online. Jonathon > pointed to this commit as problematic: > > http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=a3e95ebfc9586534fa26d7ea59f34987f6e78ad1 > > The author attribution is incorrect. I guess where there's one, there's > others too. How would we go about fixing that now? Is it worth fixing? What about a graft point between the converted and new commits? It would be transparent to users. That way the converted commits can be fixed later without having to change the new commits. -- Felipe Contreras From jpuydt at free.fr Wed Jan 21 13:31:58 2009 From: jpuydt at free.fr (Julien Puydt) Date: Wed, 21 Jan 2009 13:31:58 +0100 Subject: [gst-devel] Does Gstreamer has a element that can split one stream into two In-Reply-To: <1232529809.13038.18.camel@mini.centricular.net> References: <1232529809.13038.18.camel@mini.centricular.net> Message-ID: <497715BE.2020303@free.fr> Tim-Philipp M?ller a ?crit : > Don't forget to add a queue > after tee for each stream, e.g.: > > gst-launch-0.10 videotestsrc ! tee name=t \ > t. ! queue ! ximagesink \ > t. ! queue ! ximagesink Could you explain why? Snark on #gstreamer From t.i.m at zen.co.uk Wed Jan 21 13:17:13 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Wed, 21 Jan 2009 12:17:13 +0000 Subject: [gst-devel] GIT conversion ... DONE (hooks) In-Reply-To: <1232518526.7969.12.camel@joder> References: <1232518526.7969.12.camel@joder> Message-ID: <1232540233.22921.1.camel@mini.centricular.net> On Wed, 2009-01-21 at 07:15 +0100, Edward Hervey wrote: > The next steps are: > * Finishing/activating the post-commit hook to send mails to the > commit mailing-list > * Convert the GStreamer website pages containing cvs information to > mention the new git urls/usages. > * Update the GitMigration wiki page, and migrate some information from > that page to gstreamer/docs/ * set up pre-commit hooks? Cheers -Tim From rakesh2.sharma at aricent.com Wed Jan 21 13:45:59 2009 From: rakesh2.sharma at aricent.com (rakesh sharma) Date: Wed, 21 Jan 2009 04:45:59 -0800 (PST) Subject: [gst-devel] How to register Gstreamer Plugin with Playbin so that playbin detect my plugins In-Reply-To: <1232528128.13038.4.camel@mini.centricular.net> References: <21541362.post@talk.nabble.com> <3c1737210901191016s1848ee38q99b52ea8cec1bb2@mail.gmail.com> <21557477.post@talk.nabble.com> <1232443480.487.6.camel@mini.centricular.net> <21559965.post@talk.nabble.com> <785339900901200222g61183847q4af355dc76c1a65c@mail.gmail.com> <3c1737210901201009s219c7e52rac3a81b8056affbd@mail.gmail.com> <21578039.post@talk.nabble.com> <1232528128.13038.4.camel@mini.centricular.net> Message-ID: <21582459.post@talk.nabble.com> For decoder we are givin Codec/Decoder/Audio in GST_ELEMENT_DETAILS. If i want to do same for demux then what i have to give in GST_ELEMENT_DETAILS? -- View this message in context: http://www.nabble.com/How-to-register-Gstreamer-Plugin-with-Playbin-so-that-playbin-detect-my-plugins-tp21541362p21582459.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From t.i.m at zen.co.uk Wed Jan 21 14:03:15 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Wed, 21 Jan 2009 13:03:15 +0000 Subject: [gst-devel] Does Gstreamer has a element that can split one stream into two In-Reply-To: <497715BE.2020303@free.fr> References: <1232529809.13038.18.camel@mini.centricular.net> <497715BE.2020303@free.fr> Message-ID: <1232542995.22921.5.camel@mini.centricular.net> On Wed, 2009-01-21 at 13:31 +0100, Julien Puydt wrote: > > Don't forget to add a queue > > after tee for each stream, e.g.: > > > > gst-launch-0.10 videotestsrc ! tee name=t \ > > t. ! queue ! ximagesink \ > > t. ! queue ! ximagesink > > Could you explain why? Here's a short explanation: http://article.gmane.org/gmane.comp.video.gstreamer.devel/15904 Cheers -Tim From t.i.m at zen.co.uk Wed Jan 21 14:08:21 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Wed, 21 Jan 2009 13:08:21 +0000 Subject: [gst-devel] How to register Gstreamer Plugin with Playbin so that playbin detect my plugins In-Reply-To: <21582459.post@talk.nabble.com> References: <21541362.post@talk.nabble.com> <3c1737210901191016s1848ee38q99b52ea8cec1bb2@mail.gmail.com> <21557477.post@talk.nabble.com> <1232443480.487.6.camel@mini.centricular.net> <21559965.post@talk.nabble.com> <785339900901200222g61183847q4af355dc76c1a65c@mail.gmail.com> <3c1737210901201009s219c7e52rac3a81b8056affbd@mail.gmail.com> <21578039.post@talk.nabble.com> <1232528128.13038.4.camel@mini.centricular.net> <21582459.post@talk.nabble.com> Message-ID: <1232543301.22921.11.camel@mini.centricular.net> On Wed, 2009-01-21 at 04:45 -0800, rakesh sharma wrote: > For decoder we are givin Codec/Decoder/Audio in GST_ELEMENT_DETAILS. > If i want to do same for demux then what i have to give in > GST_ELEMENT_DETAILS? May I suggest you have a look at some of the existing demuxers and parsers in gst-inspect (or their source code)? Cheers -Tim From irfanshaikh at tataelxsi.co.in Wed Jan 21 15:56:37 2009 From: irfanshaikh at tataelxsi.co.in (irfanshaikh) Date: Wed, 21 Jan 2009 20:26:37 +0530 Subject: [gst-devel] How does an element work in pull mode ? Message-ID: <00d001c97bd8$76be1190$37033c0a@telxsi.com> Hi, Consider the following pipline, I think the elements in the following pipeline works in Push mode gst-launch filesrc location=/home/h264.dat ! video/x-h264, width=1280, height=720, framerate=30/1 ! ffmux_asf ! filesink location=/home/h264.asf What changes we have to do to make these element in pull mode ?? Regards, The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From msmith at xiph.org Wed Jan 21 19:03:53 2009 From: msmith at xiph.org (Michael Smith) Date: Wed, 21 Jan 2009 10:03:53 -0800 Subject: [gst-devel] How does an element work in pull mode ? In-Reply-To: <00d001c97bd8$76be1190$37033c0a@telxsi.com> References: <00d001c97bd8$76be1190$37033c0a@telxsi.com> Message-ID: <3c1737210901211003k14d6d5b1m4bf61d7bfc4fecd9@mail.gmail.com> On Wed, Jan 21, 2009 at 6:56 AM, irfanshaikh wrote: > Hi, > Consider the following pipline, I think the elements in the > following pipeline works in Push mode > > gst-launch filesrc location=/home/h264.dat ! video/x-h264, > width=1280, height=720, framerate=30/1 ! ffmux_asf ! filesink > location=/home/h264.asf > > What changes we have to do to make these element in pull mode ?? We very strongly recommend against use ffmux_*; they mostly don't work, and we can't help you with them. Which element do you want to run in pull mode? Presumably the muxer: you'd have to rewrite the muxer to run in pull mode. But it still won't work, since you don't have an h264 parser in your pipeline. Maybe you wanted to add a parser, then run THAT in pull mode? Either way: if it doesn't already support pull mode, you'd need to add that functionality to the source. Mike From bilboed at gmail.com Thu Jan 22 00:40:28 2009 From: bilboed at gmail.com (Edward Hervey) Date: Thu, 22 Jan 2009 00:40:28 +0100 Subject: [gst-devel] GIT conversion ... DONE In-Reply-To: <1232531364.4111.1.camel@fancy-ubuntu> References: <1232518526.7969.12.camel@joder> <1232531364.4111.1.camel@fancy-ubuntu> Message-ID: <1232581228.7969.34.camel@joder> On Wed, 2009-01-21 at 09:49 +0000, Jan Schmidt wrote: > On Wed, 2009-01-21 at 07:15 +0100, Edward Hervey wrote: > > Hello hackers, > > > > All the main GStreamer modules are now converted to git and up. > > > > You can view them here http://cgit.freedesktop.org/gstreamer/ > > The url to access each module is at the bottom of each of those > > modules. > > > > Hey hey, > > Thanks for getting the conversion done and everything online. Jonathon > pointed to this commit as problematic: > > http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=a3e95ebfc9586534fa26d7ea59f34987f6e78ad1 > > The author attribution is incorrect. I guess where there's one, there's > others too. How would we go about fixing that now? Is it worth fixing? > > J. I don't think this is a blocker issue. The authorname is still valid in the commit message, and those are *historical* commits. After discussing in #gstreamer, we can add graft points later on if we actually see actual problematic issues/commits without modifying the head commit sha1 (and therefore not screwing up everybody cloning/branching off the repositories). Edward > > > > The next steps are: > > * Finishing/activating the post-commit hook to send mails to the > > commit mailing-list > > * Convert the GStreamer website pages containing cvs information to > > mention the new git urls/usages. > > * Update the GitMigration wiki page, and migrate some information from > > that page to gstreamer/docs/ > > * Hacking > > > > I have currently got to leave for a conference dinner at LCA, but will > > be back online after that to finish those. > > > > The last step (hacking) can be done without me being online :) > > Just remember that the decision was made that we no longer commit > > changes to the ChangeLog file but only in the commit entry. The > > ChangeLog files will be automatically generated at (pre-)release time > > using a script mentionned in the GitMigration wiki page. > > > > Thanks to Daniel Stone from freedesktop.org for the help in the > > administration side of things. > > > > CVS is dead, long live GIT ! > > > > Edward > > > > > > ------------------------------------------------------------------------------ > > This SF.net email is sponsored by: > > SourcForge Community > > SourceForge wants to tell your story. > > http://p.sf.net/sfu/sf-spreadtheword > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From bilboed at gmail.com Thu Jan 22 00:55:36 2009 From: bilboed at gmail.com (Edward Hervey) Date: Thu, 22 Jan 2009 00:55:36 +0100 Subject: [gst-devel] GIT conversion ... DONE In-Reply-To: <1232581228.7969.34.camel@joder> References: <1232518526.7969.12.camel@joder> <1232531364.4111.1.camel@fancy-ubuntu> <1232581228.7969.34.camel@joder> Message-ID: <1232582136.7969.37.camel@joder> On Thu, 2009-01-22 at 00:40 +0100, Edward Hervey wrote: > On Wed, 2009-01-21 at 09:49 +0000, Jan Schmidt wrote: > > On Wed, 2009-01-21 at 07:15 +0100, Edward Hervey wrote: > > > Hello hackers, > > > > > > All the main GStreamer modules are now converted to git and up. > > > > > > You can view them here http://cgit.freedesktop.org/gstreamer/ > > > The url to access each module is at the bottom of each of those > > > modules. > > > > > > > Hey hey, > > > > Thanks for getting the conversion done and everything online. Jonathon > > pointed to this commit as problematic: > > > > http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=a3e95ebfc9586534fa26d7ea59f34987f6e78ad1 > > > > The author attribution is incorrect. I guess where there's one, there's > > others too. How would we go about fixing that now? Is it worth fixing? > > > > J. > > I don't think this is a blocker issue. The authorname is still valid > in the commit message, and those are *historical* commits. > > After discussing in #gstreamer, we can add graft points later on if we > actually see actual problematic issues/commits without modifying the > head commit sha1 (and therefore not screwing up everybody > cloning/branching off the repositories). So in fact we can't do this... because graft points are not part of the clonable data (needs to be done manually on the client-side). What I propose as an intermediate solution for this (and so we can actually get to *using* the repositories) is to push the non-malaxed/non-cleaned-up raw converted repositories (which I have) into /git/gstreamer/historical/*.git That way people can have a non-encumbered version of the repositories. Does that suit everybody ? Edward > > Edward > > > > > > > > The next steps are: > > > * Finishing/activating the post-commit hook to send mails to the > > > commit mailing-list > > > * Convert the GStreamer website pages containing cvs information to > > > mention the new git urls/usages. > > > * Update the GitMigration wiki page, and migrate some information from > > > that page to gstreamer/docs/ > > > * Hacking > > > > > > I have currently got to leave for a conference dinner at LCA, but will > > > be back online after that to finish those. > > > > > > The last step (hacking) can be done without me being online :) > > > Just remember that the decision was made that we no longer commit > > > changes to the ChangeLog file but only in the commit entry. The > > > ChangeLog files will be automatically generated at (pre-)release time > > > using a script mentionned in the GitMigration wiki page. > > > > > > Thanks to Daniel Stone from freedesktop.org for the help in the > > > administration side of things. > > > > > > CVS is dead, long live GIT ! > > > > > > Edward > > > > > > > > > ------------------------------------------------------------------------------ > > > This SF.net email is sponsored by: > > > SourcForge Community > > > SourceForge wants to tell your story. > > > http://p.sf.net/sfu/sf-spreadtheword > > > _______________________________________________ > > > gstreamer-devel mailing list > > > gstreamer-devel at lists.sourceforge.net > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > From bilboed at gmail.com Thu Jan 22 00:55:36 2009 From: bilboed at gmail.com (Edward Hervey) Date: Thu, 22 Jan 2009 00:55:36 +0100 Subject: [gst-devel] GIT conversion ... DONE In-Reply-To: <1232581228.7969.34.camel@joder> References: <1232518526.7969.12.camel@joder> <1232531364.4111.1.camel@fancy-ubuntu> <1232581228.7969.34.camel@joder> Message-ID: <1232582136.7969.37.camel@joder> On Thu, 2009-01-22 at 00:40 +0100, Edward Hervey wrote: > On Wed, 2009-01-21 at 09:49 +0000, Jan Schmidt wrote: > > On Wed, 2009-01-21 at 07:15 +0100, Edward Hervey wrote: > > > Hello hackers, > > > > > > All the main GStreamer modules are now converted to git and up. > > > > > > You can view them here http://cgit.freedesktop.org/gstreamer/ > > > The url to access each module is at the bottom of each of those > > > modules. > > > > > > > Hey hey, > > > > Thanks for getting the conversion done and everything online. Jonathon > > pointed to this commit as problematic: > > > > http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=a3e95ebfc9586534fa26d7ea59f34987f6e78ad1 > > > > The author attribution is incorrect. I guess where there's one, there's > > others too. How would we go about fixing that now? Is it worth fixing? > > > > J. > > I don't think this is a blocker issue. The authorname is still valid > in the commit message, and those are *historical* commits. > > After discussing in #gstreamer, we can add graft points later on if we > actually see actual problematic issues/commits without modifying the > head commit sha1 (and therefore not screwing up everybody > cloning/branching off the repositories). So in fact we can't do this... because graft points are not part of the clonable data (needs to be done manually on the client-side). What I propose as an intermediate solution for this (and so we can actually get to *using* the repositories) is to push the non-malaxed/non-cleaned-up raw converted repositories (which I have) into /git/gstreamer/historical/*.git That way people can have a non-encumbered version of the repositories. Does that suit everybody ? Edward > > Edward > > > > > > > > The next steps are: > > > * Finishing/activating the post-commit hook to send mails to the > > > commit mailing-list > > > * Convert the GStreamer website pages containing cvs information to > > > mention the new git urls/usages. > > > * Update the GitMigration wiki page, and migrate some information from > > > that page to gstreamer/docs/ > > > * Hacking > > > > > > I have currently got to leave for a conference dinner at LCA, but will > > > be back online after that to finish those. > > > > > > The last step (hacking) can be done without me being online :) > > > Just remember that the decision was made that we no longer commit > > > changes to the ChangeLog file but only in the commit entry. The > > > ChangeLog files will be automatically generated at (pre-)release time > > > using a script mentionned in the GitMigration wiki page. > > > > > > Thanks to Daniel Stone from freedesktop.org for the help in the > > > administration side of things. > > > > > > CVS is dead, long live GIT ! > > > > > > Edward > > > > > > > > > ------------------------------------------------------------------------------ > > > This SF.net email is sponsored by: > > > SourcForge Community > > > SourceForge wants to tell your story. > > > http://p.sf.net/sfu/sf-spreadtheword > > > _______________________________________________ > > > gstreamer-devel mailing list > > > gstreamer-devel at lists.sourceforge.net > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > From weian.chen at intel.com Thu Jan 22 03:00:42 2009 From: weian.chen at intel.com (Chen, Weian) Date: Thu, 22 Jan 2009 10:00:42 +0800 Subject: [gst-devel] Can application get the dynamic information inside the element Message-ID: Hi all: If the application wants to get the dynamic information inside an element (such as the file handle opened by filesrc in runtime), how to do? Thanks in advanced. Weian -------------- next part -------------- An HTML attachment was scrubbed... URL: From msmith at xiph.org Thu Jan 22 03:18:15 2009 From: msmith at xiph.org (Michael Smith) Date: Wed, 21 Jan 2009 18:18:15 -0800 Subject: [gst-devel] Can application get the dynamic information inside the element In-Reply-To: References: Message-ID: <3c1737210901211818k5248bed0i261fb6d3d50f85dd@mail.gmail.com> On Wed, Jan 21, 2009 at 6:00 PM, Chen, Weian wrote: > Hi all: > > If the application wants to get the dynamic information inside an element > (such as the file handle opened by filesrc in runtime), how to do? The element may expose things via properties. For your specific question: filesrc does expose the fd it's using (via an 'fd' property), but it's almost certainly a bad idea to use that for any purpose. Why do you want to know that? Mike From weian.chen at intel.com Thu Jan 22 03:54:34 2009 From: weian.chen at intel.com (Chen, Weian) Date: Thu, 22 Jan 2009 10:54:34 +0800 Subject: [gst-devel] Can application get the dynamic information inside the element In-Reply-To: <3c1737210901211818k5248bed0i261fb6d3d50f85dd@mail.gmail.com> References: <3c1737210901211818k5248bed0i261fb6d3d50f85dd@mail.gmail.com> Message-ID: Thanks you for your info. My usage is to open the device in source element, and I want to control it in my application, but I can not open the device twice. So what's your suggestion? Weian -----Original Message----- From: Michael Smith [mailto:msmith at xiph.org] Sent: 2009?1?22? 10:18 To: Discussion of the development of GStreamer Subject: Re: [gst-devel] Can application get the dynamic information inside the element On Wed, Jan 21, 2009 at 6:00 PM, Chen, Weian wrote: > Hi all: > > If the application wants to get the dynamic information inside an element > (such as the file handle opened by filesrc in runtime), how to do? The element may expose things via properties. For your specific question: filesrc does expose the fd it's using (via an 'fd' property), but it's almost certainly a bad idea to use that for any purpose. Why do you want to know that? Mike ------------------------------------------------------------------------------ This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From irfanshaikh at tataelxsi.co.in Thu Jan 22 05:37:02 2009 From: irfanshaikh at tataelxsi.co.in (irfanshaikh) Date: Thu, 22 Jan 2009 10:07:02 +0530 Subject: [gst-devel] How does an element work in pull mode ? In-Reply-To: <3c1737210901211003k14d6d5b1m4bf61d7bfc4fecd9@mail.gmail.com> Message-ID: <00db01c97c4b$12bd3130$37033c0a@telxsi.com> Hi, Thanks for your reply !!! I have modified the pipeline with the h264parse element. gst-launch filesrc location=/home/h264.dat ! h264parse ! video/x-h264, width=1280, height=720, framerate=30/1 ! ffmux_asf ! filesink location=/home/h264.asf Is it ok if the h264parse work in pull mode and ffmux_asf work in push mode in the pipeline ?? Please tell me how do i make h264parse to work in pull mode ? If you suggest me how to make an element to work in pull mode. I can try the rest of elements in the pipeline to work in pull mode(also ffmux_asf). Any small Hint would be helpful !! Thanks in advance !!!! regards, -----Original Message----- From: mlrsmith at gmail.com [mailto:mlrsmith at gmail.com]On Behalf Of Michael Smith Sent: Wednesday, January 21, 2009 11:34 PM To: irfanshaikh at tataelxsi.co.in; Discussion of the development of GStreamer Subject: Re: [gst-devel] How does an element work in pull mode ? On Wed, Jan 21, 2009 at 6:56 AM, irfanshaikh wrote: > Hi, > Consider the following pipline, I think the elements in the > following pipeline works in Push mode > > gst-launch filesrc location=/home/h264.dat ! video/x-h264, > width=1280, height=720, framerate=30/1 ! ffmux_asf ! filesink > location=/home/h264.asf > > What changes we have to do to make these element in pull mode ?? We very strongly recommend against use ffmux_*; they mostly don't work, and we can't help you with them. Which element do you want to run in pull mode? Presumably the muxer: you'd have to rewrite the muxer to run in pull mode. But it still won't work, since you don't have an h264 parser in your pipeline. Maybe you wanted to add a parser, then run THAT in pull mode? Either way: if it doesn't already support pull mode, you'd need to add that functionality to the source. Mike The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From bilboed at gmail.com Thu Jan 22 05:54:57 2009 From: bilboed at gmail.com (Edward Hervey) Date: Thu, 22 Jan 2009 05:54:57 +0100 Subject: [gst-devel] GIT conversion ... DONE (hooks) In-Reply-To: <1232540233.22921.1.camel@mini.centricular.net> References: <1232518526.7969.12.camel@joder> <1232540233.22921.1.camel@mini.centricular.net> Message-ID: <1232600097.4556.1.camel@joder> On Wed, 2009-01-21 at 12:17 +0000, Tim-Philipp M?ller wrote: > On Wed, 2009-01-21 at 07:15 +0100, Edward Hervey wrote: > > > The next steps are: > > * Finishing/activating the post-commit hook to send mails to the > > commit mailing-list DONE > > * Convert the GStreamer website pages containing cvs information to > > mention the new git urls/usages. DONE > > * Update the GitMigration wiki page, and migrate some information from > > that page to gstreamer/docs/ The website 'developers' page has been updated to point to the new repositories. > > * set up pre-commit hooks? DONE (automatically installed by autogen.sh). I'm currently updating each module, but it's taking some time (slow internet here). > > Cheers > -Tim > > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From rakesh2.sharma at aricent.com Thu Jan 22 08:21:30 2009 From: rakesh2.sharma at aricent.com (rakesh sharma) Date: Wed, 21 Jan 2009 23:21:30 -0800 (PST) Subject: [gst-devel] How to register Gstreamer Plugin with Playbin so that playbin detect my plugins In-Reply-To: <1232543301.22921.11.camel@mini.centricular.net> References: <21541362.post@talk.nabble.com> <3c1737210901191016s1848ee38q99b52ea8cec1bb2@mail.gmail.com> <21557477.post@talk.nabble.com> <1232443480.487.6.camel@mini.centricular.net> <21559965.post@talk.nabble.com> <785339900901200222g61183847q4af355dc76c1a65c@mail.gmail.com> <3c1737210901201009s219c7e52rac3a81b8056affbd@mail.gmail.com> <21578039.post@talk.nabble.com> <1232528128.13038.4.camel@mini.centricular.net> <21582459.post@talk.nabble.com> <1232543301.22921.11.camel@mini.centricular.net> Message-ID: <21599063.post@talk.nabble.com> Is there any way to know which elements are used by playbin...... I want to know which demux is used by decodebin. -- View this message in context: http://www.nabble.com/How-to-register-Gstreamer-Plugin-with-Playbin-so-that-playbin-detect-my-plugins-tp21541362p21599063.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From listjiason at gmail.com Thu Jan 22 09:14:10 2009 From: listjiason at gmail.com (Jiason Li) Date: Thu, 22 Jan 2009 16:14:10 +0800 Subject: [gst-devel] codec benchmarking Message-ID: Hi, I'm new to the list, thus any suggestions are welcomed. :D What I want to do is to perform a benchmark on a specific codec plugin, like "mplayer -benchmark", to get info about average cpu usage, resource consumption, etc., but I could not find a way and have no idea how to do so with gstreamer. please give me some advice (methodology/tools/etc.), truly thanks. :) sincerely, jiason 20090122 From vaisakh.n at nestgroup.net Thu Jan 22 10:26:19 2009 From: vaisakh.n at nestgroup.net (vaisakh.n) Date: Thu, 22 Jan 2009 14:56:19 +0530 Subject: [gst-devel] gstreamer-data flow Message-ID: <49783BBB.4090903@nestgroup.net> How is the data flow among the elements in gstreamer pipeline is it pass full buffer among the elements? ***** Confidentiality Statement/Disclaimer ***** This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. From sebastian.droege at collabora.co.uk Thu Jan 22 10:38:28 2009 From: sebastian.droege at collabora.co.uk (Sebastian =?ISO-8859-1?Q?Dr=F6ge?=) Date: Thu, 22 Jan 2009 10:38:28 +0100 Subject: [gst-devel] GIT conversion ... DONE (hooks) In-Reply-To: <1232600097.4556.1.camel@joder> References: <1232518526.7969.12.camel@joder> <1232540233.22921.1.camel@mini.centricular.net> <1232600097.4556.1.camel@joder> Message-ID: <1232617108.561.15.camel@odin.lan> Am Donnerstag, den 22.01.2009, 05:54 +0100 schrieb Edward Hervey: > > * set up pre-commit hooks? > DONE (automatically installed by autogen.sh). I'm currently updating > each module, but it's taking some time (slow internet here). The current pre-commit hook is broken if files are moved/deleted. The patch below fixes this. --- .git/hooks/pre-commit 2009-01-22 09:33:28.000000000 +0100 +++ .git/hooks/pre-commit 2009-01-22 10:36:36.000000000 +0100 @@ -7,7 +7,8 @@ echo "--Checking style--" for file in `git-diff-index --name-only HEAD | grep "\.c$"` ; do -tempfoo=`basename $0` + test -f ${file} || continue + tempfoo=`basename $0` newfile=`mktemp /tmp/${tempfoo}.XXXXXX` || exit 1 indent \ --braces-on-if-line \ -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: Dies ist ein digital signierter Nachrichtenteil URL: From sebastian.droege at collabora.co.uk Thu Jan 22 11:28:45 2009 From: sebastian.droege at collabora.co.uk (Sebastian =?ISO-8859-1?Q?Dr=F6ge?=) Date: Thu, 22 Jan 2009 11:28:45 +0100 Subject: [gst-devel] GIT conversion ... DONE (hooks) In-Reply-To: <1232617108.561.15.camel@odin.lan> References: <1232518526.7969.12.camel@joder> <1232540233.22921.1.camel@mini.centricular.net> <1232600097.4556.1.camel@joder> <1232617108.561.15.camel@odin.lan> Message-ID: <1232620125.561.21.camel@odin.lan> Am Donnerstag, den 22.01.2009, 10:38 +0100 schrieb Sebastian Dr?ge: > Am Donnerstag, den 22.01.2009, 05:54 +0100 schrieb Edward Hervey: > > > * set up pre-commit hooks? > > DONE (automatically installed by autogen.sh). I'm currently updating > > each module, but it's taking some time (slow internet here). > > The current pre-commit hook is broken if files are moved/deleted. The > patch below fixes this. And this still causes problems because indent is not giving the same result when called twice on a file. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: Dies ist ein digital signierter Nachrichtenteil URL: From maximlevitsky at gmail.com Thu Jan 22 12:10:46 2009 From: maximlevitsky at gmail.com (Maxim Levitsky) Date: Thu, 22 Jan 2009 13:10:46 +0200 Subject: [gst-devel] Few questions about status of gstreamer In-Reply-To: <1232490570.15685.35.camel@maxim-laptop> References: <1232490570.15685.35.camel@maxim-laptop> Message-ID: <1232622646.5197.0.camel@maxim-laptop> On Wed, 2009-01-21 at 00:29 +0200, Maxim Levitsky wrote: > Hi, > > I was tinkering with gstreamer lately, and I find it qute nice platform. > > I would like to know where current development is headed, and what to > expect. > > While the concept of gstreamer is great, the current implementation > isn't that great, I would like to point at following things: > > - web live streams playback is very bad compared even to other linux > media players, as a test case visit any web site that offers list of > live web channels, and you will be able to watch maybe 8% og them or so, > this is very sad. I know that there are property features, but many > streams are yet playable in vlc or mplayer. > > I wish there were a general database of links to such streams that will > be visited by gstreamer/totem developers as this will encourage them to > fix support for that. > Also such database will ease the regression testing. > ( for example ms switched to asf over rtsp, and I see some streams that > use it, but gstreamer will just fail as it will think it is a mms stream > also support for wmv over rtsp isn't in gstreamer, patches are aviable, > but it doesn't seem that they will be mereged soon. sad. I have no free > time yet ( might have some in near future ) to do that myself. > > ( note that I am speaking about web sites that offer list of live > streams, as they provide direct links without fancy media player > testing, this rules out web page incompatibilities due to sites using > windows only player controls) > > > - gstreamer elements are far from beeng generic: > look at that for example : > > maxim at maxim-laptop:~$ gst-inspect xvidenc | grep video > Long name: XviD video encoder > video/x-raw-yuv > video/x-raw-rgb > video/x-raw-rgb > video/x-raw-rgb > video/x-raw-rgb > video/x-raw-rgb > video/x-raw-rgb > video/x-raw-rgb > video/x-xvid > > as you see it supports only 'video/x-xvid' on output, but why, isn't > xvid output plain mpeg-4 ? > > on the other hand > > maxim at maxim-laptop:~$ gst-inspect ffmux_mp4 | grep video > video/quicktime > SINK template: 'video_%d' > video/mpeg > video/x-divx > video/x-h264 > > this is just one of may examples. > > in other words currently muxers are very picky about formats they can > handle although underlying formats can hold much more codecs > AVI for example can pretty much hold any codec that have fourcc code. > > > also note that sinks of muxers sometimes are named video_[0-9] and > sometimes video_[0-9][0-9] > > while it is possible to avoid referencing those names directly, still it > would be better to have standard naming > > > - There are many pipeline stalls. > it might be no joy to sit for a 4 hours to figure out > 'where did I forgot to put that 'queue' > > also without magic 'videorate' that drops frames on demand (right?) it > is very hard/impossible to create working pipeline, (not to mention that > you absolutely need queues after each encoder) > > also generally there is no way to copy a raw stream from one container > to another. > > I for example tried to put asf stream (captured into file) ( gstream > doesn't seem to support seeking in it) to avi, which is possible and is > done nicely by mencoder. > > Gstreamer just stalls. > > I figured out that the problem was that intially input stream contained > one big (~ 27 KB) chunk of video data, and then audio, but this made > asfdemux stall, as it can't pull both video and audio data seperatly > > In other words demuxers can easily stall if input stream isn't evenly > interleaved. > > > > > Best regards, > Maxim Levitsky > > PS: do you know whenever gstreamer can add an index to avi file? > > I use this command to record a live asf stream: > > > > #source description > > export SOURCE_PROTOCOL=mmssrc > > export SOURCE_DEMUXER=asfdemux > > export SOURCE_AUDIO_CODEC=ffdec_wmav2 > > export SOURCE_VIDEO_CODEC=ffdec_wmv3 > > > > #destanation description > > export DEST_PROTOCOL=filesink > > export DEST_MUXER=avimux > > export DEST_AUDIO_CODEC=lame > > export DEST_VIDEO_CODEC=xvidenc > > > gst-launch \ > > $SOURCE_PROTOCOL location=$SOURCE_LOCATION ! queue ! $SOURCE_DEMUXER name=demuxer \ > > \ > > demuxer. ! queue ! $SOURCE_VIDEO_CODEC ! videorate ! postproc_default ! tee name=raw_video \ > > demuxer. ! queue ! $SOURCE_AUDIO_CODEC ! tee name=raw_audio \ > > \ > > raw_video. ! queue leaky=1 ! $DEST_VIDEO_CODEC ! queue ! $DEST_MUXER name=muxer \ > > raw_audio. ! queue leaky=1 ! $DEST_AUDIO_CODEC ! queue ! muxer. \ > > \ > > raw_video. ! queue ! gconfvideosink \ > > raw_audio. ! queue ! gconfaudiosink \ > > \ > > muxer. ! $DEST_PROTOCOL location=$DEST_LOCATION \ > > It works, but resulting file can be seeking to non key-frames, which > obliviously make seeking very unpleasant. > (since muxers are limited to such small number of codecs, can't they > know which frames are key and which not ?) > > > Hi, What do you think? Maxim Levitsky From amm at tid.es Thu Jan 22 15:06:01 2009 From: amm at tid.es (Antonio =?ISO-8859-1?Q?Marqu=E9s?=) Date: Thu, 22 Jan 2009 15:06:01 +0100 Subject: [gst-devel] plugin working with realimentation In-Reply-To: <496F2615.8010904@hora-obscura.de> References: <1231931318.5877.742.camel@viladecans> <496F2615.8010904@hora-obscura.de> Message-ID: <1232633161.5877.1016.camel@viladecans> Sorry, what I meant is that I want to have as an input of a plugin the output of other plugin that comes after this one. It is a kind of feedback. The first plugin delivers a buffer, which is processed by another filter. The buffer pushed by this filter would be also pushed to the previous plugin. I have tried to do so in Gstreamer and it did not work at all. I had this output: 0:00:02.508437079 28071 0x6060b0 WARN bin gstbin.c:1747:gst_bin_sort_iterator_next: loop dected in graph (gst-launch-0.10:28071): GStreamer-WARNING **: loop detected in the graph of bin pipeline0!! I would like to know if there is a way in implementing this feedback from a downstream element to an upstream element. Sorry and thanks in advanced On Thu, 2009-01-15 at 13:03 +0100, Stefan Kost wrote: > hi, > Antonio Marqu?s schrieb: > > Hello developers, > > > > I am interesting to design a plugin which has as inputs N-1 number of > > pads (some images from a filesrc for example) and another input which > > would be the result of the plugin's output, so I would like to implement > > a kind of realimentation in that plugin. > > > honestly, I did not understood what you want to do. Also > "realimentation" seem to be a strange word (seems to be a medical term). > If you want to have a history of the last N buffers you need to > implement a ringbuffer that would keep a reference of them. > > Stefan > > In that plugin I would like to have all the N input buffers in the > > process function. So I have thought in using the collect pad method for > > all the pads. The problem is that, for the first buffer I would have not > > the buffer of the pad N (because it would be the output of the buffer), > > so the pipeline would not be set on PLAYING state never. Does anyone > > know how to solve this problem? The main purpose of the plugin is, to > > make an initial computation for the first frame taking into account the > > N-1 images and for the other frames taking also into account the > > previous output, which would be the N pad. > > > > If anyone knows a plugin with similar function it would be great for me > > to look at its source code. > > > > Thanks in advance. > > > > > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -- Toni Marqu?s Marqu?s Telef?nica I+D Divisi?n de Tecnolog?as de Video amm at tid.es 933653188 From jack.foster11 at gmail.com Thu Jan 22 15:30:46 2009 From: jack.foster11 at gmail.com (Jack Foster) Date: Thu, 22 Jan 2009 09:30:46 -0500 Subject: [gst-devel] Grabbing RTSP Stats? In-Reply-To: References: Message-ID: Thanks for the reply! On Wed, Jan 14, 2009 at 2:36 AM, Eric Zhang wrote: > 1. RTCP does the communication works between server & clients. Also > gstrtpbin handles the RTCP stuffs. Both my sender and receiver are using gstrtpbin. The pads the sender are rtcp rx/tx and rtp tx. Similarly the pads the receiver is using are rtcp rx/tx and rtcp rx. I guess my question specifically is, can I get access to the packets statistics on both the sender's and receiver's rtcp pads. This will provide me with the feedback of how many packets are being received vs being sent. I'm hoping to give the user feedback like at the bottom of the ekiga SIP client. Example: http://wiki.ekiga.org/index.php/Image:Ekiga_in_a_Call.png > 2. gstrtpjitterbuffer will send a customized event to downstream > elements when it finds packet missing. You can make a packet receiving > report according to this. I want to know the information even if a packet isn't missing. i.e. informing the user that all packets are being received! Thanks, Jevin > > Eric Zhang > > 2009/1/13 Jack Foster >> >> Hiya Gang, >> >> I currently have an rtp/rtsp server -> client stream via gstrtpbin. >> Is there a way I can get access to the statistics of the sender and >> receiver reports? I'd like the server to know if the client missed >> packets or if the communication path is fully broken. Would this be >> the on-timeout signal? >> >> Jack >> >> >> ------------------------------------------------------------------------------ >> This SF.net email is sponsored by: >> SourcForge Community >> SourceForge wants to tell your story. >> http://p.sf.net/sfu/sf-spreadtheword >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From rmcouat at smartt.com Thu Jan 22 18:03:51 2009 From: rmcouat at smartt.com (Ron McOuat) Date: Thu, 22 Jan 2009 09:03:51 -0800 Subject: [gst-devel] plugin working with realimentation In-Reply-To: <1232633161.5877.1016.camel@viladecans> References: <1231931318.5877.742.camel@viladecans> <496F2615.8010904@hora-obscura.de> <1232633161.5877.1016.camel@viladecans> Message-ID: <4978A6F7.5000604@smartt.com> It would likely be easier to combine the 2 elements you are considering. Then from the outside it would have only a sink pad for input and a source pad for output. Inside the combined element you would have a queue of N buffers where the first N-1 are a chain of the last N-1 buffers received on the sink and buffer N is a copy of the output sent on the source after processing of the N-1 input buffers. The processing algorithm can then process the N buffers by referencing their data as they sit in the queue. The buffer sequence for the N-1 inputs you could model on what the queue element does as a starting point for implementation. That is what I believe Stefan had in mind when he said use a ring buffer. The assumption in all of this is the first N-1 buffers are simply the N-1 most recent buffers from the input unaltered and only buffer N provides the feedback. You may have problems with pre-roll during pipeline startup, a buffer needs to get from the initial source to the final sink and it sounds like you need to process N-1 input buffers before your first output on this element. Hope that helps. Ron Antonio Marqu?s wrote: > Sorry, what I meant is that I want to have as an input of a plugin the > output of other plugin that comes after this one. It is a kind of > feedback. > > The first plugin delivers a buffer, which is processed by another > filter. The buffer pushed by this filter would be also pushed to the > previous plugin. I have tried to do so in Gstreamer and it did not work > at all. I had this output: > > > 0:00:02.508437079 28071 0x6060b0 WARN bin > gstbin.c:1747:gst_bin_sort_iterator_next: loop dected in > graph > > (gst-launch-0.10:28071): GStreamer-WARNING **: loop detected in the > graph of bin pipeline0!! > > I would like to know if there is a way in implementing this feedback > from a downstream element to an upstream element. > > Sorry and thanks in advanced > On Thu, 2009-01-15 at 13:03 +0100, Stefan Kost wrote: > >> hi, >> Antonio Marqu?s schrieb: >> >>> Hello developers, >>> >>> I am interesting to design a plugin which has as inputs N-1 number of >>> pads (some images from a filesrc for example) and another input which >>> would be the result of the plugin's output, so I would like to implement >>> a kind of realimentation in that plugin. >>> >>> >> honestly, I did not understood what you want to do. Also >> "realimentation" seem to be a strange word (seems to be a medical term). >> If you want to have a history of the last N buffers you need to >> implement a ringbuffer that would keep a reference of them. >> >> Stefan >> >>> In that plugin I would like to have all the N input buffers in the >>> process function. So I have thought in using the collect pad method for >>> all the pads. The problem is that, for the first buffer I would have not >>> the buffer of the pad N (because it would be the output of the buffer), >>> so the pipeline would not be set on PLAYING state never. Does anyone >>> know how to solve this problem? The main purpose of the plugin is, to >>> make an initial computation for the first frame taking into account the >>> N-1 images and for the other frames taking also into account the >>> previous output, which would be the N pad. >>> >>> If anyone knows a plugin with similar function it would be great for me >>> to look at its source code. >>> >>> Thanks in advance. >>> >>> >>> >> ------------------------------------------------------------------------------ >> This SF.net email is sponsored by: >> SourcForge Community >> SourceForge wants to tell your story. >> http://p.sf.net/sfu/sf-spreadtheword >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> From msmith at xiph.org Thu Jan 22 18:58:59 2009 From: msmith at xiph.org (Michael Smith) Date: Thu, 22 Jan 2009 09:58:59 -0800 Subject: [gst-devel] gstreamer-data flow In-Reply-To: <49783BBB.4090903@nestgroup.net> References: <49783BBB.4090903@nestgroup.net> Message-ID: <3c1737210901220958k24ca7245x8a5e2211f152ec91@mail.gmail.com> On Thu, Jan 22, 2009 at 1:26 AM, vaisakh.n wrote: > How is the data flow among the elements in gstreamer pipeline > > is it pass full buffer among the elements? You can find the answer to this, and many more questions, in the fine manual we wrote for your benefit. It contains extensive information on how gstreamer pipelines work. Whilst we're happy to answer questions on this list on complex topics, simple questions that indicate you haven't put any effort into reading the existing documentation are less likely to get a helpful response. Mike From thaytan at noraisin.net Thu Jan 22 23:47:46 2009 From: thaytan at noraisin.net (thaytan at noraisin.net) Date: Thu, 22 Jan 2009 22:47:46 +0000 (GMT) Subject: [gst-devel] RELEASE: GStreamer Good Plug-ins 0.10.13 'Blatant Discouragement' Message-ID: <20090122224746.3237344126@for.noraisin.net> This mail announces the release of GStreamer Good Plug-ins 0.10.13 'Blatant Discouragement'. GStreamer Good Plug-ins is a set of plug-ins that we consider to have good quality code and correct functionality, under our preferred license (LGPL for the plug-in code, LGPL or LGPL-compatible for the supporting library). For more information, see http://gstreamer.freedesktop.org/modules/gst-plugins-good.html To file bugs, go to http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gst-plugins-good -------------- next part -------------- Release notes for GStreamer Good Plug-ins??0.10.13 "Blatant Discouragement" The GStreamer team is proud to announce a new release in the 0.10.x stable series of the GStreamer Good Plug-ins. The 0.10.x series is a stable series targeted at end users. It is not API or ABI compatible with the stable 0.8.x series. It is, however, parallel installable with the 0.8.x series. "Such ingratitude. After all the times I've saved your life." A collection of plug-ins you'd want to have right next to you on the battlefield. Shooting sharp and making no mistakes, these plug-ins have it all: good looks, good code, and good licensing. Documented and dressed up in tests. If you're looking for a role model to base your own plug-in on, here it is. If you find a plot hole or a badly lip-synced line of code in them, let us know - it is a matter of honour for us to ensure Blondie doesn't look like he's been walking 100 miles through the desert without water. This module contains a set of plug-ins that we consider to have good quality code, correct functionality, our preferred license (LGPL for the plug-in code, LGPL or LGPL-compatible for the supporting library). We believe distributors can safely ship these plug-ins. People writing elements should base their code on these elements. Other modules containing plug-ins are: gst-plugins-base contains a basic set of well-supported plug-ins gst-plugins-ugly contains a set of well-supported plug-ins, but might pose problems for distributors gst-plugins-bad contains a set of less supported plug-ins that haven't passed the rigorous quality testing we expect Features of this release * Fix for security advisory TKADV2009-0xx: Fix potential buffer overflows while reading quicktime headers. * Fix bad autopoint substitution in the po subdir There were no bugs fixed in this release Download You can find source releases of gst-plugins-good in the download directory: http://gstreamer.freedesktop.org/src/gst-plugins-good/ GStreamer Homepage More details can be found on the project's website: http://gstreamer.freedesktop.org/ Support and Bugs We use GNOME's bugzilla for bug reports and feature requests: http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer Developers CVS is hosted on cvs.freedesktop.org. All code is in CVS and can be checked out from there. Interested developers of the core library, plug-ins, and applications should subscribe to the gstreamer-devel list. If there is sufficient interest we will create more lists as necessary. Applications Contributors to this release * David Schleef * Jan Schmidt ?? From bilboed at gmail.com Fri Jan 23 00:20:30 2009 From: bilboed at gmail.com (Edward Hervey) Date: Fri, 23 Jan 2009 00:20:30 +0100 Subject: [gst-devel] Fwd: GStreamer subtitles task In-Reply-To: <34539a480901200925l56ed1fdexd1ee34a84808679e@mail.gmail.com> References: <537e39c90901200906t2d351e31ndc9f73a7659a17cc@mail.gmail.com> <34539a480901200925l56ed1fdexd1ee34a84808679e@mail.gmail.com> Message-ID: <1232666430.31046.5.camel@joder> > ---------- Forwarded message ---------- > From: Jordi Jaen Pallares > Date: Tue, Jan 20, 2009 at 12:06 PM > Subject: GStreamer subtitles task > To: thomas at apestaart.org, rbultje at ronald.bitfreak.net > > > Hi ! > > do you have someone to help you with the subtitles issue in GStreamer ? > > I am working for an european project (http://www.ict-ccast.eu/) that > will need subtitle support. > Since I am also pushing GStreamer as a principal streaming engine in > the project, I would like to contribute to the subtitles task as > announced in > http://gstreamer.freedesktop.org/tasks/gstreamer.html > > Please let me know if u still need help on that and what is required. > > I would be interested in some kind of plug-in to blend some text as > subtitles on a video. Subtitle support is split up in two types of elements: * textoverlay, which takes both a video and a text input stream and outputs a video stream with the text rendered over it * an element providing a timed-text stream. Some demuxers support this (if you have subtitles in a Video file for example), and there are also some standalone elements to parse separate subtitle formats (see the subparse element for such an element). The task page is clearly outdated :( Edward > > Best regards, > > Jordi > > > PS. some time ago I sent a small patch for the GStreamer MPEG-TS RTP > packetizer plugin > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From bilboed at gmail.com Fri Jan 23 00:20:30 2009 From: bilboed at gmail.com (Edward Hervey) Date: Fri, 23 Jan 2009 00:20:30 +0100 Subject: [gst-devel] Fwd: GStreamer subtitles task In-Reply-To: <34539a480901200925l56ed1fdexd1ee34a84808679e@mail.gmail.com> References: <537e39c90901200906t2d351e31ndc9f73a7659a17cc@mail.gmail.com> <34539a480901200925l56ed1fdexd1ee34a84808679e@mail.gmail.com> Message-ID: <1232666430.31046.5.camel@joder> > ---------- Forwarded message ---------- > From: Jordi Jaen Pallares > Date: Tue, Jan 20, 2009 at 12:06 PM > Subject: GStreamer subtitles task > To: thomas at apestaart.org, rbultje at ronald.bitfreak.net > > > Hi ! > > do you have someone to help you with the subtitles issue in GStreamer ? > > I am working for an european project (http://www.ict-ccast.eu/) that > will need subtitle support. > Since I am also pushing GStreamer as a principal streaming engine in > the project, I would like to contribute to the subtitles task as > announced in > http://gstreamer.freedesktop.org/tasks/gstreamer.html > > Please let me know if u still need help on that and what is required. > > I would be interested in some kind of plug-in to blend some text as > subtitles on a video. Subtitle support is split up in two types of elements: * textoverlay, which takes both a video and a text input stream and outputs a video stream with the text rendered over it * an element providing a timed-text stream. Some demuxers support this (if you have subtitles in a Video file for example), and there are also some standalone elements to parse separate subtitle formats (see the subparse element for such an element). The task page is clearly outdated :( Edward > > Best regards, > > Jordi > > > PS. some time ago I sent a small patch for the GStreamer MPEG-TS RTP > packetizer plugin > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From dtaylor at startrac.com Fri Jan 23 00:55:37 2009 From: dtaylor at startrac.com (Dan Taylor) Date: Thu, 22 Jan 2009 15:55:37 -0800 Subject: [gst-devel] How to register Gstreamer Plugin withPlaybin so that playbin detect my plugins In-Reply-To: <21599063.post@talk.nabble.com> References: <21541362.post@talk.nabble.com><3c1737210901191016s1848ee38q99b52ea8cec1bb2@mail.gmail.com><21557477.post@talk.nabble.com><1232443480.487.6.camel@mini.centricular.net><21559965.post@talk.nabble.com><785339900901200222g61183847q4af355dc76c1a65c@mail.gmail.com><3c1737210901201009s219c7e52rac3a81b8056affbd@mail.gmail.com><21578039.post@talk.nabble.com><1232528128.13038.4.camel@mini.centricular.net><21582459.post@talk.nabble.com><1232543301.22921.11.camel@mini.centricular.net> <21599063.post@talk.nabble.com> Message-ID: <1822849CB0478545ADCFB217EF4A340503F967BD@sedah.startrac.com> > -----Original Message----- > From: rakesh sharma [mailto:rakesh2.sharma at aricent.com] > Sent: Wednesday, January 21, 2009 11:22 PM > To: gstreamer-devel at lists.sourceforge.net > Subject: Re: [gst-devel] How to register Gstreamer Plugin withPlaybin so > that playbin detect my plugins > > > Is there any way to know which elements are used by playbin...... > I want to know which demux is used by decodebin. If you use the "-m" option, you will see messages from the elements' state changes, and can get the elements from there. From bilboed at gmail.com Fri Jan 23 05:53:18 2009 From: bilboed at gmail.com (Edward Hervey) Date: Fri, 23 Jan 2009 05:53:18 +0100 Subject: [gst-devel] add a plugin In-Reply-To: <818500.46623.qm@web94809.mail.in2.yahoo.com> References: <49756471.60105@nestgroup.net> <818500.46623.qm@web94809.mail.in2.yahoo.com> Message-ID: <1232686398.4100.1.camel@joder> On Tue, 2009-01-20 at 11:36 +0530, sai charmboy wrote: > go to hell. dont send messages again I beg your pardon ? Can someone remove that person (charmboy) from the mailing list ? It's what he wants after all. Edward > > saicharmboy... > > > > > ______________________________________________________________________ > From: vaisakh.n > To: Discussion of the development of GStreamer > > Sent: Tuesday, 20 January, 2009 11:13:13 AM > Subject: [gst-devel] add a plugin > > Hi, > I have a decoder plugin's library file ( .so file ) > how can i add it to gstreamer plugin registry.. > please help me > what are the steps i need to follow? > advance thanks > > ***** Confidentiality Statement/Disclaimer ***** > > This message and any attachments is intended for the sole use of the > intended recipient. It may contain confidential information. Any > unauthorized use, dissemination or modification is strictly > prohibited. If you are not the intended recipient, please notify the > sender immediately then delete it from all your systems, and do not > copy, use or print. Internet communications are not secure and it is > the responsibility of the recipient to make sure that it is > virus/malicious code exempt. > The company/sender cannot be responsible for any unauthorized > alterations or modifications made to the contents. If you require any > form of confirmation of the contents, please contact the > company/sender. The company/sender is not liable for any errors or > omissions in the content of this message. > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists..sourceforge.net/lists/listinfo/gstreamer-devel > > > > ______________________________________________________________________ > Check out the all-new face of Yahoo! India. Click here. > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From rahul.verma at sasken.com Fri Jan 23 12:46:43 2009 From: rahul.verma at sasken.com (Rahul Verma) Date: Fri, 23 Jan 2009 17:16:43 +0530 Subject: [gst-devel] 3gp demux plugin working on PC but not on target In-Reply-To: References: Message-ID: Hi, >ffmpeg_demux should not be used as a reference plugin for anything. >It's not a good example to use at all. Don't use it for any >comparisons; the ffmpeg demuxers are known to work poorly (and in many >cases not work at all). >I don't actually know what it does internally - I haven't looked - >but a gstreamer demuxer plugin can't act on a file directly, because >it doesn't even know if there IS a file involved at all. >By "it must pull data" I mean that to get data from the upstream >element (in pull mode), you must use gst_pad_pull_range(). Thanks Mike, this helped a lot. I was referring to ffmpeg_demux all this while and was going on the wrong track. As you said, I used gst_pad_pull_range() and it works wonderfully well. Now I don't have any dependency on the file source file handle. One additional thing I needed to clarify. I tried using the same buffer returned by gst_pad_pull_range() in the function gst_pad_push() and got a crash. I tried functions like gst_buffer_make_writeable(), but even that did not help much. For the time being I am doing a memcpy and pushing the data, which I think is not a clean way. It is also causing audio breaks. Is there a way I can bypass this memcpy?? >I don't know either. Sorry, I can only give general suggestions, since >you're not writing an open source plugin. Looking at the debug output >(GST_DEBUG) might help you. This AV sync is the last major issue I am facing. For a few streams, AV sync works fine, but for one particular stream I am having a pretty hard time. Anyways, I will keep my experiments going and try to keep you informed too. GST_DEBUG is also of not much help because GST_DEBUG:3 and above causes crash at random places when I run the code on target. Regards, Rahul Verma SASKEN BUSINESS DISCLAIMER ------------------------- This message may contain confidential, proprietary or legally privileged information. 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However the company accepts no liability for any damage caused by any virus transmitted by this email From ensonic at hora-obscura.de Fri Jan 23 23:13:39 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sat, 24 Jan 2009 00:13:39 +0200 Subject: [gst-devel] Can application get the dynamic information inside the element In-Reply-To: References: <3c1737210901211818k5248bed0i261fb6d3d50f85dd@mail.gmail.com> Message-ID: <497A4113.8020806@hora-obscura.de> Chen, Weian schrieb: > Thanks you for your info. > > My usage is to open the device in source element, and I want to control it in my application, but I can not open the device twice. > > So what's your suggestion? > > Weian > -----Original Message----- > From: Michael Smith [mailto:msmith at xiph.org] > Sent: 2009?1?22? 10:18 > To: Discussion of the development of GStreamer > Subject: Re: [gst-devel] Can application get the dynamic information inside the element > > On Wed, Jan 21, 2009 at 6:00 PM, Chen, Weian wrote: >> Hi all: >> >> If the application wants to get the dynamic information inside an element >> (such as the file handle opened by filesrc in runtime), how to do? > Thats exactly when a file-descriptor 'fd' property can be used. Just look at fdsrc element in core. Stefan > The element may expose things via properties. > > For your specific question: filesrc does expose the fd it's using (via > an 'fd' property), but it's almost certainly a bad idea to use that > for any purpose. Why do you want to know that? > > Mike > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From ensonic at hora-obscura.de Fri Jan 23 23:09:29 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sat, 24 Jan 2009 00:09:29 +0200 Subject: [gst-devel] How to register Gstreamer Plugin with Playbin so that playbin detect my plugins In-Reply-To: <21599063.post@talk.nabble.com> References: <21541362.post@talk.nabble.com> <3c1737210901191016s1848ee38q99b52ea8cec1bb2@mail.gmail.com> <21557477.post@talk.nabble.com> <1232443480.487.6.camel@mini.centricular.net> <21559965.post@talk.nabble.com> <785339900901200222g61183847q4af355dc76c1a65c@mail.gmail.com> <3c1737210901201009s219c7e52rac3a81b8056affbd@mail.gmail.com> <21578039.post@talk.nabble.com> <1232528128.13038.4.camel@mini.centricular.net> <21582459.post@talk.nabble.com> <1232543301.22921.11.camel@mini.centricular.net> <21599063.post@talk.nabble.com> Message-ID: <497A4019.1030205@hora-obscura.de> rakesh sharma schrieb: > Is there any way to know which elements are used by playbin...... > I want to know which demux is used by decodebin. use -v for gst-launch Stefan From gstreamer at tech-no-logical.net Fri Jan 23 22:43:28 2009 From: gstreamer at tech-no-logical.net (gstreamer at tech-no-logical.net) Date: Fri, 23 Jan 2009 22:43:28 +0100 Subject: [gst-devel] gstreamer as a streamtranscoder Message-ID: <20090123214328.GB2324@tech-no-logical.net> hi. i'm trying to use gstreamer to transcode an ogg-stream to an mp3-stream, both running on the same icecast host. seems simple : uridecodebin ! lame ! shout2send and done (with or without some ! queue elements in there somewhere). i can't seem to get it to work though. whether using neonhttpsrc, gnomevfssrc or uridecodebin, whenever a meta-data update is received, the element involved sees an eos-message and dies : > gst-launch -vvv neonhttpsrc location="http://localhost:8000/test.ogg" ! queue ! oggdemux ! vorbisdec ! audioconvert ! lame ! queue ! shout2send ip=127.0.0.10.1 port=8000 password=hackme mount=mp3 Setting pipeline to PAUSED ... Pipeline is PREROLLING ... /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:src: caps = audio/x-raw-float, rate=(int)44100, channels=(int)2, endianness=(int)1234, width=(int)32 /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:sink: caps = audio/x-vorbis /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = audio/x-raw-float, rate=(int)44100, channels=(int)2, endianness=(int)1234, width=(int)32 /GstPipeline:pipeline0/GstLame:lame0.GstPad:src: caps = audio/mpeg, mpegversion=(int)1, mpegaudioversion=(int)1, layer=(int)3, channels=(int)2, rate=(int)44100 /GstPipeline:pipeline0/GstLame:lame0.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 /GstPipeline:pipeline0/GstQueue:queue1.GstPad:sink: caps = audio/mpeg, mpegversion=(int)1, mpegaudioversion=(int)1, layer=(int)3, channels=(int)2, rate=(int)44100 /GstPipeline:pipeline0/GstQueue:queue1.GstPad:src: caps = audio/mpeg, mpegversion=(int)1, mpegaudioversion=(int)1, layer=(int)3, channels=(int)2, rate=(int)44100 /GstPipeline:pipeline0/GstShout2send:shout2send0.GstPad:sink: caps = audio/mpeg, mpegversion=(int)1, mpegaudioversion=(int)1, layer=(int)3, channels=(int)2, rate=(int)44100 Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstSystemClock Got EOS from element "pipeline0". Execution ended after 34083719000 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... /GstPipeline:pipeline0/GstShout2send:shout2send0.GstPad:sink: caps = NULL /GstPipeline:pipeline0/GstQueue:queue1.GstPad:src: caps = NULL /GstPipeline:pipeline0/GstQueue:queue1.GstPad:sink: caps = NULL /GstPipeline:pipeline0/GstLame:lame0.GstPad:src: caps = NULL /GstPipeline:pipeline0/GstLame:lame0.GstPad:sink: caps = NULL /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps = NULL /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = NULL /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:src: caps = NULL /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:sink: caps = NULL /GstPipeline:pipeline0/GstOggDemux:oggdemux0.GstOggPad:serial_64880eeb: caps = NULL Setting pipeline to NULL ... FREEING pipeline ... or, with uridecodebin : gst-launch -vvv uridecodebin uri=http://localhost:8000/test.ogg ! audioconvert ! lame ! shout2send ip=127.0.0.1 port=8000 password=hackme mount=mp3 Setting pipeline to PAUSED ... /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0: source = (GstGnomeVFSSrc) source /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstGnomeVFSSrc:source: iradio-genre = "Various" /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstGnomeVFSSrc:source: iradio-name = "My Station name" /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstGnomeVFSSrc:source: iradio-url = "http://www.audiorealm.com" Pipeline is PREROLLING ... /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstTypeFindElement:typefindelement0.GstPad:src: caps = application/ogg /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20/GstTypeFindElement:typefind: force-caps = application/ogg /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20: sink-caps = application/ogg /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20/GstTypeFindElement:typefind.GstPad:src: caps = application/ogg /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstQueue2:queue20.GstPad:sink: caps = application/ogg /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstQueue2:queue20.GstPad:src: caps = application/ogg /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20/GstTypeFindElement:typefind.GstPad:sink: caps = application/ogg /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20.GstGhostPad:sink: caps = application/ogg /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20.GstGhostPad:sink: caps = application/ogg /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20.GstGhostPad:sink.GstProxyPad:proxypad0: caps = application/ogg /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20/GstOggDemux:oggdemux0.GstPad:sink: caps = application/ogg /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20/GstMultiQueue:multiqueue0.GstPad:sink0: caps = audio/x-vorbis /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20/GstMultiQueue:multiqueue0: max-size-buffers = 5 /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20/GstMultiQueue:multiqueue0: max-size-time = 2000000000 /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20/GstMultiQueue:multiqueue0: max-size-bytes = 2097152 /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20/GstMultiQueue:multiqueue0.GstPad:src0: caps = audio/x-vorbis /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20/GstVorbisDec:vorbisdec1.GstPad:sink: caps = audio/x-vorbis /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = audio/x-raw-float, rate=(int)44100, channels=(int)2, endianness=(int)1234, width=(int)32 /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0.GstGhostPad:src0: caps = audio/x-raw-float, rate=(int)44100, channels=(int)2, endianness=(int)1234, width=(int)32 /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0.GstGhostPad:src0.GstProxyPad:proxypad2: caps = audio/x-raw-float, rate=(int)44100, channels=(int)2, endianness=(int)1234, width=(int)32 /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20.GstGhostPad:src0: caps = audio/x-raw-float, rate=(int)44100, channels=(int)2, endianness=(int)1234, width=(int)32 /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20.GstGhostPad:src0.GstProxyPad:proxypad1: caps = audio/x-raw-float, rate=(int)44100, channels=(int)2, endianness=(int)1234, width=(int)32 /GstPipeline:pipeline0/GstLame:lame0.GstPad:src: caps = audio/mpeg, mpegversion=(int)1, mpegaudioversion=(int)1, layer=(int)3, channels=(int)2, rate=(int)44100 /GstPipeline:pipeline0/GstLame:lame0.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 /GstPipeline:pipeline0/GstShout2send:shout2send0.GstPad:sink: caps = audio/mpeg, mpegversion=(int)1, mpegaudioversion=(int)1, layer=(int)3, channels=(int)2, rate=(int)44100 Prerolled, waiting for buffering to finish... Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstSystemClock /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20/GstMultiQueue:multiqueue1.GstPad:sink0: caps = audio/x-vorbis /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20/GstVorbisDec:vorbisdec3.GstPad:src: caps = audio/x-raw-float, rate=(int)44100, channels=(int)2, endianness=(int)1234, width=(int)32 /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20/GstMultiQueue:multiqueue1: max-size-buffers = 5 /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20/GstMultiQueue:multiqueue1: max-size-time = 2000000000 /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20/GstMultiQueue:multiqueue1: max-size-bytes = 2097152 /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20/GstMultiQueue:multiqueue1: max-size-buffers = 5 /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20/GstMultiQueue:multiqueue1: max-size-time = 2000000000 /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20/GstMultiQueue:multiqueue1: max-size-bytes = 2097152 /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20/GstMultiQueue:multiqueue1.GstPad:src0: caps = audio/x-vorbis /GstPipeline:pipeline0/GstURIDecodeBin:uridecodebin0/GstDecodeBin2:decodebin20/GstVorbisDec:vorbisdec3.GstPad:sink: caps = audio/x-vorbis Buffering, setting pipeline to PAUSED ... at which point it just sits there and waits. it all works flawlessly until the next track-change on the icecast side... some more info : openbsd 4.4 gstreamer-0.10.21 gst-plugins-base-0.10.21 gst-plugins-good-0.10.11 gst-plugins-bad-0.10.9 gst-plugins-ugly-0.10.10 icecast-2.3.1 any thoughts ? -- CUL8R, Peter. From hayden at heydan.co.uk Sat Jan 24 09:58:42 2009 From: hayden at heydan.co.uk (Hayden Andrews) Date: Sat, 24 Jan 2009 15:58:42 +0700 Subject: [gst-devel] Saving a live stream Message-ID: <497AD842.8040706@heydan.co.uk> Hi guys, We have a live A/V Ogg stream that runs for a couple of hours, but we want to be able to save an (approximately) 40 min section of this stream to a file. First of all, I wrote a quick Perl script which reads directly from the appropriate HTTP port (an Icecast Server) and writes to a file. Everything looks fine with this file except .. 1) Suppose we record from 40 mins to 1:20 of the live stream. When we play the file, the current time starts at the time we started recording from rather than from 00:00. Although the run time is correct (40 mins), the current time progresses from 40mins to 1:20. 2) When someone tries to edit the file to trim each end and make any other minor changes that are required, seeking takes forever and the editor takes for ever to do things that would normally be quite quick with a normal file. I believe that if the file has a play time of about 40 mins, then the player says that it will take about 40 mins to do many tasks. I figure that since a live stream has an unknown length, and since it is not possible to seek when viewing a live stream, that there is missing info in the live stream that needs to be inserted while saving. So, I also figure that it would be easy to insert the relevant requirements while saving a live stream to file..... So, I edited my script to use Gstreamer instead. I have used souphttpsrc ! filesink and I've used souphttpsrc ! decodebin ! ...... ! oggmux ! filesink. I hoped that the filesink would expect to insert [what ever is needed] and would do [something] automatically. But we still get files which are difficult to edit. So, I tried setting do-timestamp=true in southttpsrc which produced a file, which when played by vlc, showed either a very short (2-3 seconds) or unknown run time, and crashed the player when I tried seeking within the file being played. So, what should I be doing? Any ideas.... I expect that this is a simple and normal scenario as I'm sure that it is more than normal to be saving portions of live streams to file for later editing/playing. Cheers, Hayden From ensonic at hora-obscura.de Sat Jan 24 17:13:40 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sat, 24 Jan 2009 18:13:40 +0200 Subject: [gst-devel] Building failing for qtwrapper In-Reply-To: References: Message-ID: <497B3E34.6050604@hora-obscura.de> thiagoss schrieb: > Qtwraper building is failing (cvs head), I don't know if this is a > misconfiguration in my system or in Makefile, so I'm sending this mail. > The error output is: > > Making all in qtwrapper > cd ../.. && /bin/sh > /Users/thiagoss/Work/GStreamer/head/gst-plugins-bad/missing --run > automake-1.10 --gnu sys/qtwrapper/Makefile > configure.ac:111 : warning: > AC_CACHE_VAL(g_support_dowhile_macros, ...): suspicious cache-id, must > contain _cv_ to be cached > ../../lib/autoconf/general.m4:1974: AC_CACHE_VAL is expanded from... > ../../lib/autoconf/general.m4:1994: AC_CACHE_CHECK is expanded from... > common/m4/gst-dowhile.m4:8: AG_GST_CHECK_DOWHILE_MACROS is expanded from... > configure.ac:111 : the top level > configure.ac:143 : warning: > AC_CACHE_VAL(GST_FIONREAD_IN_SYS_IOCTL, ...): suspicious cache-id, must > contain _cv_ to be cached > m4/gst-fionread.m4:1: GST_CHECK_FIONREAD is expanded from... > configure.ac:143 : the top level > configure.ac:143 : warning: > AC_CACHE_VAL(GST_FIONREAD_IN_SYS_FILIO, ...): suspicious cache-id, must > contain _cv_ to be cached > cd ../.. && /bin/sh ./config.status sys/qtwrapper/Makefile depfiles > config.status: creating sys/qtwrapper/Makefile > config.status: executing depfiles commands > /bin/sh ../../libtool --tag=CC --tag=disable-static --mode=compile gcc > -DHAVE_CONFIG_H -I. -I../.. -I../../gst-libs -I../../gst-libs > -D_REENTRANT -I/usr/local/include/gstreamer-0.10 > -I/opt/local/include/glib-2.0 -I/opt/local/lib/glib-2.0/include > -I/opt/local/include -I/opt/local/include/libxml2 -Wall > -Wdeclaration-after-statement -Werror -fno-strict-aliasing -g > -D_REENTRANT -I/usr/local/include/gstreamer-0.10 > -I/opt/local/include/glib-2.0 -I/opt/local/lib/glib-2.0/include > -I/opt/local/include -I/opt/local/include/libxml2 -g -O2 -MT > libgstqtwrapper_la-qtwrapper.lo -MD -MP -MF > .deps/libgstqtwrapper_la-qtwrapper.Tpo -c -o > libgstqtwrapper_la-qtwrapper.lo `test -f 'qtwrapper.c' || echo > './'`qtwrapper.c > libtool: compile: gcc -DHAVE_CONFIG_H -I. -I../.. -I../../gst-libs > -I../../gst-libs -D_REENTRANT -I/usr/local/include/gstreamer-0.10 > -I/opt/local/include/glib-2.0 -I/opt/local/lib/glib-2.0/include > -I/opt/local/include -I/opt/local/include/libxml2 -Wall > -Wdeclaration-after-statement -Werror -fno-strict-aliasing -g > -D_REENTRANT -I/usr/local/include/gstreamer-0.10 > -I/opt/local/include/glib-2.0 -I/opt/local/lib/glib-2.0/include > -I/opt/local/include -I/opt/local/include/libxml2 -g -O2 -MT > libgstqtwrapper_la-qtwrapper.lo -MD -MP -MF > .deps/libgstqtwrapper_la-qtwrapper.Tpo -c qtwrapper.c -fno-common -DPIC > -o .libs/libgstqtwrapper_la-qtwrapper.o > In file included from qtwrapper.c:49: > qtwrapper.h:53:20: error: Movies.h: No such file or directory > qtwrapper.c: In function 'plugin_init': > qtwrapper.c:58: error: 'OSErr' undeclared (first use in this function) > qtwrapper.c:58: error: (Each undeclared identifier is reported only once > qtwrapper.c:58: error: for each function it appears in.) > qtwrapper.c:58: error: syntax error before 'status' > qtwrapper.c:69: error: 'status' undeclared (first use in this function) > cc1: warnings being treated as errors > qtwrapper.c:69: warning: implicit declaration of function 'EnterMovies' > make[3]: *** [libgstqtwrapper_la-qtwrapper.lo] Error 1 > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > How to fix this? It seems that configure belives you have all the needed external libs and headers, but in fact you don't. Do you have a Movies.h? Maybe you can look at config.log and check what its looking for. Stefan > > -- > Thiaogo Sousa Santos > > Embedded Systems and Pervasive Computing Lab (Embedded) > Center of Electrical Engineering and Informatics (CEEI) > Federal University of Campina Grande (UFCG) > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > > > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From ensonic at hora-obscura.de Sun Jan 25 16:20:38 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sun, 25 Jan 2009 17:20:38 +0200 Subject: [gst-devel] installation of gstreamer on windows In-Reply-To: References: <20875253.post@talk.nabble.com> <392293.71799.qm@web36508.mail.mud.yahoo.com> Message-ID: <497C8346.6000100@hora-obscura.de> sumit kumar schrieb: > i have tried version 0.10.5 and 0.10.6, but it dint help.. please guide.. x264enc is under gst-plugins-bad/ext, that means it uses an external library for the encoding. You need to also build and install x264enc to get the gstreamer plugin for it built. Stefan > > On Wed, Dec 10, 2008 at 9:37 PM, sumit kumar > wrote: > > which version is needed for that? thanks in advance.. > :) > -sumit > > On Wed, Dec 10, 2008 at 8:17 PM, Bill Cholewka > wrote: > > install gst-plugins-bad > > ------------------------------------------------------------------------ > *From:* sumit kumar > > *To:* Discussion of the development of GStreamer > > > *Sent:* Tuesday, December 9, 2008 8:45:22 AM > *Subject:* Re: [gst-devel] installation of gstreamer on windows > > Hi, > I have installed the gstreamer using OABuild.I am trying to > stream h264 raw video data, but it requires a plugin called > 'x264enc' which is not there by default. Does anyone knows > encoder plugin available for H264 on windows. > > -Sumit > > On Sun, Dec 7, 2008 at 3:48 AM, Thomas Loewe > wrote: > > > Hi, > > i have cross compiled gst-ffmpeg-0.10.6 under xubuntu 8.10, > but there's > nothing ffmpeg related with gst-inspect to find after > installing it under > windows xp. > > Are there any dependencies or something i forgot? > > Thanks, > Thomas > -- > View this message in context: > http://www.nabble.com/installation-of-gstreamer-on-windows-tp20836599p20875253.html > Sent from the GStreamer-devel mailing list archive at > Nabble.com . > > > ------------------------------------------------------------------------------ > SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las > Vegas, Nevada. > The future of the web can't happen without you. Join us at > MIX09 to help > pave the way to the Next Web now. Learn more and register at > http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las > Vegas, Nevada. > The future of the web can't happen without you. Join us at > MIX09 to help > pave the way to the Next Web now. Learn more and register at > http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------------ > SF.Net email is Sponsored by MIX09, March 18-20, 2009 in Las Vegas, Nevada. > The future of the web can't happen without you. Join us at MIX09 to help > pave the way to the Next Web now. Learn more and register at > http://ad.doubleclick.net/clk;208669438;13503038;i?http://2009.visitmix.com/ > > > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From ylatuya at gmail.com Sun Jan 25 19:58:30 2009 From: ylatuya at gmail.com (Andoni Morales) Date: Sun, 25 Jan 2009 19:58:30 +0100 Subject: [gst-devel] Error using x264 encoder plugin in Windows Message-ID: <772db3280901251058i1c91321atafdaad8315a6aa4@mail.gmail.com> Hi, I've testing the x264 encoder plugin in Windows (gst-plugins-bad 0.10.10) and I always get an error complaining about no ratecontrol method specified. The pipeline used is: gst-launch-0.10.exe -v videotestsrc ! x264enc ! avimux ! filesink location=t.avi And the error: 0:00:00.312500000 4068 01AABE30 ERROR x264enc .:0:: no ratecontrol method specified 0:00:00.312500000 4068 01AABE30 WARN default .:0:: error: Can not initialize x264 encoder. 0:00:00.312500000 4068 01AABE30 INFO GST_ERROR_SYSTEM .:0:: posting message: Can not initialize x264 encoder. 0:00:00.312500000 4068 01AABE30 INFO GST_ERROR_SYSTEM .:0:: posting message: Can not initialize x264 encoder. 0:00:00.312500000 4068 01AABE30 INFO GST_ERROR_SYSTEM .:0:: posted error message: Can not initialize x264 encoder. ERROR: from element /GstPipeline:pipeline0/GstX264Enc:x264enc0: Can not initialize x264 encoder. Additional debug info: ..\..\ext\x264\gstx264enc.c(620): gst_x264_enc_init_encoder (): /GstPipeline:pipeline0/GstX264Enc:x264enc0 Does any one knows what can be wrong? -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilboed at gmail.com Mon Jan 26 10:36:51 2009 From: bilboed at gmail.com (Edward Hervey) Date: Mon, 26 Jan 2009 10:36:51 +0100 Subject: [gst-devel] [gst-cvs] 'Good' GStreamer plugins branch, pulse-bg567794, created. RELEASE-0_10_11-164-gebe6eac In-Reply-To: <20090125023958.8200510096@kemper.freedesktop.org> References: <20090125023958.8200510096@kemper.freedesktop.org> Message-ID: <1232962611.3853.1.camel@localhost> Hi Jan, Although tempting, I don't think it's wise to have work branches on the official repositories. Please use a personal repository on fdo for that. Edward On Sat, 2009-01-24 at 18:39 -0800, Jan Schmidt wrote: > This is an automated email from the git hooks/post-receive script. It was > generated because a ref change was pushed to the repository containing > the project "'Good' GStreamer plugins". > > The branch, pulse-bg567794 has been created > at ebe6eacfef8d09278e5508c9f13d0fff92e4f967 (commit) > > - Log ----------------------------------------------------------------- > commit ebe6eacfef8d09278e5508c9f13d0fff92e4f967 > Author: Jan Schmidt > Date: Sun Jan 25 02:35:57 2009 +0000 > > Rework pulse plugin. > > Attempt to merge changes for bug 567794 with changes since > gst-plugins-good 0.9.11, and add conditional defines to allow > compilation with older pulseaudio installations. > > ----------------------------------------------------------------------- > > > hooks/post-receive > -- > 'Good' GStreamer plugins > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-cvs mailing list > gstreamer-cvs at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-cvs From msmith at xiph.org Mon Jan 26 19:32:52 2009 From: msmith at xiph.org (Michael Smith) Date: Mon, 26 Jan 2009 10:32:52 -0800 Subject: [gst-devel] 3gp demux plugin working on PC but not on target In-Reply-To: References: Message-ID: <3c1737210901261032j7bcec835r23d88f6ef690a97@mail.gmail.com> On Fri, Jan 23, 2009 at 3:46 AM, Rahul Verma wrote: > > Thanks Mike, this helped a lot. I was referring to ffmpeg_demux all this while and was going on the wrong track. As you said, I used gst_pad_pull_range() and it works wonderfully well. Now I don't have any dependency on the file source file handle. > > One additional thing I needed to clarify. I tried using the same buffer returned by gst_pad_pull_range() in the function gst_pad_push() and got a crash. I tried functions like gst_buffer_make_writeable(), but even that did not help much. For the time being I am doing a memcpy and pushing the data, which I think is not a clean way. It is also causing audio breaks. Is there a way I can bypass this memcpy?? You should be able to push a buffer that you have pulled from upstream. If you can't, then presumably you're getting your refcounting wrong somewhere (or there's some other problem; hard to say...). When you just say "got a crash" it's not too helpful - you need to at least find out where it's crashing, and why. > >>I don't know either. Sorry, I can only give general suggestions, since >>you're not writing an open source plugin. Looking at the debug output >>(GST_DEBUG) might help you. > > This AV sync is the last major issue I am facing. For a few streams, AV sync works fine, but for one particular stream I am having a pretty hard time. Anyways, I will keep my experiments going and try to keep you informed too. GST_DEBUG is also of not much help because GST_DEBUG:3 and above causes crash at random places when I run the code on target. Well, that seems like a much more serious problem that you should fix first! If GST_DEBUG isn't working, then there's something seriously wrong with (probably) lower-level software that gstreamer relies on. Possibly you've compiled glib incorrectly or something. If you don't fix that, then you shouldn't expect gstreamer to work correctly at all - this could easily be the cause of your other crashes. Mike From ensonic at hora-obscura.de Mon Jan 26 23:13:13 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Tue, 27 Jan 2009 00:13:13 +0200 Subject: [gst-devel] Few questions about status of gstreamer In-Reply-To: <1232490570.15685.35.camel@maxim-laptop> References: <1232490570.15685.35.camel@maxim-laptop> Message-ID: <497E3579.1070808@hora-obscura.de> Maxim Levitsky schrieb: > Hi, > > I was tinkering with gstreamer lately, and I find it qute nice platform. > > I would like to know where current development is headed, and what to > expect. > > While the concept of gstreamer is great, the current implementation > isn't that great, I would like to point at following things: > > - web live streams playback is very bad compared even to other linux > media players, as a test case visit any web site that offers list of > live web channels, and you will be able to watch maybe 8% og them or so, > this is very sad. I know that there are property features, but many > streams are yet playable in vlc or mplayer. I don't think its that bad, but please file bugs. Thats the only way to improve it. Its helpful if you can try to narrow down whats causing the failure, like if you can cluster the features that cause problems. > > I wish there were a general database of links to such streams that will > be visited by gstreamer/totem developers as this will encourage them to > fix support for that. > Also such database will ease the regression testing. > ( for example ms switched to asf over rtsp, and I see some streams that > use it, but gstreamer will just fail as it will think it is a mms stream > also support for wmv over rtsp isn't in gstreamer, patches are aviable, > but it doesn't seem that they will be mereged soon. sad. I have no free > time yet ( might have some in near future ) to do that myself. > > ( note that I am speaking about web sites that offer list of live > streams, as they provide direct links without fancy media player > testing, this rules out web page incompatibilities due to sites using > windows only player controls) > > > - gstreamer elements are far from beeng generic: > look at that for example : > > maxim at maxim-laptop:~$ gst-inspect xvidenc | grep video > Long name: XviD video encoder > video/x-raw-yuv > video/x-raw-rgb > video/x-raw-rgb > video/x-raw-rgb > video/x-raw-rgb > video/x-raw-rgb > video/x-raw-rgb > video/x-raw-rgb > video/x-xvid > > as you see it supports only 'video/x-xvid' on output, but why, isn't > xvid output plain mpeg-4 ? > > on the other hand > > maxim at maxim-laptop:~$ gst-inspect ffmux_mp4 | grep video > video/quicktime > SINK template: 'video_%d' > video/mpeg > video/x-divx > video/x-h264 > > this is just one of may examples. Again please file bugs. You are right we have issues like this. The sheer amount of plugins requires quite some effort on maintenance. Of course bug with patches are even nicer. > > in other words currently muxers are very picky about formats they can > handle although underlying formats can hold much more codecs > AVI for example can pretty much hold any codec that have fourcc code. > > > also note that sinks of muxers sometimes are named video_[0-9] and > sometimes video_[0-9][0-9] > > while it is possible to avoid referencing those names directly, still it > would be better to have standard naming > Unfortunately this can't be changed in the stable development series :/ > > - There are many pipeline stalls. > it might be no joy to sit for a 4 hours to figure out > 'where did I forgot to put that 'queue' > Yes thats annoying, but imho its not easy to detect this from within gstreamer. If you know a way, eternal fame awaits you. > > also without magic 'videorate' that drops frames on demand (right?) it > is very hard/impossible to create working pipeline, (not to mention that > you absolutely need queues after each encoder) > I can't follow here. Many elements implement qos and they will take messures on high cpu load like dropping frames. The queues are a nice concept for multithreading. > > also generally there is no way to copy a raw stream from one container > to another. Thats wrong. > > I for example tried to put asf stream (captured into file) ( gstream > doesn't seem to support seeking in it) to avi, which is possible and is > done nicely by mencoder. Of course you can't put an asf into an avi. Both are containers. Besides seeking in asf works, seeking on mms streams very recently got implemented. > > Gstreamer just stalls. Tell us what pipelines you have used and we might be able to help. > > I figured out that the problem was that intially input stream contained > one big (~ 27 KB) chunk of video data, and then audio, but this made > asfdemux stall, as it can't pull both video and audio data seperatly > > In other words demuxers can easily stall if input stream isn't evenly > interleaved. Yes, use big enough queues after the demuxer, or even better use multiqueue. > > > Best regards, > Maxim Levitsky > > PS: do you know whenever gstreamer can add an index to avi file? avimux could take a round of improvements. Not sure about that. long mail = lots of answers. lets look at the issues one by one Stefan > > I use this command to record a live asf stream: > > >> #source description >> export SOURCE_PROTOCOL=mmssrc >> export SOURCE_DEMUXER=asfdemux >> export SOURCE_AUDIO_CODEC=ffdec_wmav2 >> export SOURCE_VIDEO_CODEC=ffdec_wmv3 >> >> #destanation description >> export DEST_PROTOCOL=filesink >> export DEST_MUXER=avimux >> export DEST_AUDIO_CODEC=lame >> export DEST_VIDEO_CODEC=xvidenc > >> gst-launch \ >> $SOURCE_PROTOCOL location=$SOURCE_LOCATION ! queue ! $SOURCE_DEMUXER name=demuxer \ >> \ >> demuxer. ! queue ! $SOURCE_VIDEO_CODEC ! videorate ! postproc_default ! tee name=raw_video \ >> demuxer. ! queue ! $SOURCE_AUDIO_CODEC ! tee name=raw_audio \ >> \ >> raw_video. ! queue leaky=1 ! $DEST_VIDEO_CODEC ! queue ! $DEST_MUXER name=muxer \ >> raw_audio. ! queue leaky=1 ! $DEST_AUDIO_CODEC ! queue ! muxer. \ >> \ >> raw_video. ! queue ! gconfvideosink \ >> raw_audio. ! queue ! gconfaudiosink \ >> \ >> muxer. ! $DEST_PROTOCOL location=$DEST_LOCATION \ > > It works, but resulting file can be seeking to non key-frames, which > obliviously make seeking very unpleasant. > (since muxers are limited to such small number of codecs, can't they > know which frames are key and which not ?) > > > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From erappleman at gmail.com Tue Jan 27 00:24:11 2009 From: erappleman at gmail.com (Eric Appleman) Date: Mon, 26 Jan 2009 18:24:11 -0500 Subject: [gst-devel] Implementing gstassrender Message-ID: <497E461B.80302@gmail.com> Would it be possible to evaluate and include Martin Sourada's new plugin to parse and render libass-compatible styled subtitles? http://bugzilla.gnome.org/show_bug.cgi?id=481075#c23 Thanks in advance. - Eric From ylatuya at gmail.com Tue Jan 27 00:33:05 2009 From: ylatuya at gmail.com (Andoni Morales) Date: Tue, 27 Jan 2009 00:33:05 +0100 Subject: [gst-devel] Few questions about status of gstreamer In-Reply-To: <1232490570.15685.35.camel@maxim-laptop> References: <1232490570.15685.35.camel@maxim-laptop> Message-ID: <772db3280901261533i2433fe69y5737fc4519ee7c65@mail.gmail.com> > > PS: do you know whenever gstreamer can add an index to avi file? > > I use this command to record a live asf stream: > > > > #source description > > export SOURCE_PROTOCOL=mmssrc > > export SOURCE_DEMUXER=asfdemux > > export SOURCE_AUDIO_CODEC=ffdec_wmav2 > > export SOURCE_VIDEO_CODEC=ffdec_wmv3 > > > > #destanation description > > export DEST_PROTOCOL=filesink > > export DEST_MUXER=avimux > > export DEST_AUDIO_CODEC=lame > > export DEST_VIDEO_CODEC=xvidenc > > > gst-launch > \ > > $SOURCE_PROTOCOL location=$SOURCE_LOCATION ! queue ! > $SOURCE_DEMUXER name=demuxer \ > > > \ > > demuxer. ! queue ! $SOURCE_VIDEO_CODEC ! videorate ! > postproc_default ! tee name=raw_video \ > > demuxer. ! queue ! $SOURCE_AUDIO_CODEC ! tee name=raw_audio > \ > > > \ > > raw_video. ! queue leaky=1 ! $DEST_VIDEO_CODEC ! queue ! > $DEST_MUXER name=muxer \ > > raw_audio. ! queue leaky=1 ! $DEST_AUDIO_CODEC ! queue ! muxer. > \ > > > \ > > raw_video. ! queue ! gconfvideosink > \ > > raw_audio. ! queue ! gconfaudiosink > \ > > > \ > > muxer. ! $DEST_PROTOCOL location=$DEST_LOCATION > \ > > It works, but resulting file can be seeking to non key-frames, which > obliviously make seeking very unpleasant. > (since muxers are limited to such small number of codecs, can't they > know which frames are key and which not ?) You just have to fire an EOS event to finalize the file and build the index -------------- next part -------------- An HTML attachment was scrubbed... URL: From rakesh2.sharma at aricent.com Tue Jan 27 12:18:25 2009 From: rakesh2.sharma at aricent.com (rakesh sharma) Date: Tue, 27 Jan 2009 03:18:25 -0800 (PST) Subject: [gst-devel] How to register Gstreamer Plugin with Playbin so that playbin detect my plugins In-Reply-To: <497A4019.1030205@hora-obscura.de> References: <21541362.post@talk.nabble.com> <3c1737210901191016s1848ee38q99b52ea8cec1bb2@mail.gmail.com> <21557477.post@talk.nabble.com> <1232443480.487.6.camel@mini.centricular.net> <21559965.post@talk.nabble.com> <785339900901200222g61183847q4af355dc76c1a65c@mail.gmail.com> <3c1737210901201009s219c7e52rac3a81b8056affbd@mail.gmail.com> <21578039.post@talk.nabble.com> <1232528128.13038.4.camel@mini.centricular.net> <21582459.post@talk.nabble.com> <1232543301.22921.11.camel@mini.centricular.net> <21599063.post@talk.nabble.com> <497A4019.1030205@hora-obscura.de> Message-ID: <21683555.post@talk.nabble.com> Can anybody tell me what this function do in depth... gst_element_make_from_uri() the description is like this : Creates an element for handling the given URI If i want that it will use my own created plugin is it possible ??? My plugin dont have the uri property, whether i have to add uri property and do some more changes ??? -- View this message in context: http://www.nabble.com/How-to-register-Gstreamer-Plugin-with-Playbin-so-that-playbin-detect-my-plugins-tp21541362p21683555.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From bilboed at gmail.com Tue Jan 27 13:38:40 2009 From: bilboed at gmail.com (Edward Hervey) Date: Tue, 27 Jan 2009 13:38:40 +0100 Subject: [gst-devel] Demux In-Reply-To: References: Message-ID: <1233059920.3853.4.camel@localhost> On Mon, 2009-01-19 at 20:18 +0530, Krishna Kishore wrote: > Hi, > > I have seen that everyone uses opensource ffmpeg demux or qtdemux. > I did not come across anyone writing their own demux using their own > parser module. > > Why does everyone use this opensource demux? Is it because it is > already available, well tested? > OR did anyone try it and faced problems and switched to opensource > demux? ... I think the *real* reason why everybody does that is because it's pointless/insane/masochistic to recode things that are already available. If there's something you want that qtdemux doesn't implement yet, file a bug asking for that. Edward > > > Regards, > Kishore. > SASKEN BUSINESS DISCLAIMER > ------------------------- > This message may contain confidential, proprietary or legally privileged information. In > case you are not the original intended Recipient of the message, you must not, directly or > indirectly, use, Disclose, distribute, print, or copy any part of this message and you are > requested to delete it and inform the sender. Any views expressed in this message are > those of the individual sender unless otherwise stated. Nothing contained in this message > shall be construed as an offer or acceptance of any offer by Sasken Communication > Technologies Limited ("Sasken") unless sent with that express intent and with due > authority of Sasken. Sasken has taken enough precautions to prevent the spread of > viruses. However the company accepts no liability for any damage caused by any virus > transmitted by this email Mails where the disclaimer is bigger than the actual message normally don't get answered. > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From rahul.verma at sasken.com Tue Jan 27 14:46:42 2009 From: rahul.verma at sasken.com (Rahul Verma) Date: Tue, 27 Jan 2009 19:16:42 +0530 Subject: [gst-devel] 3gp demux plugin working on PC but not on target In-Reply-To: References: Message-ID: Hi, >You should be able to push a buffer that you have pulled from >upstream. If you can't, then presumably you're getting your >refcounting wrong somewhere (or there's some other problem; hard to >say...). When you just say "got a crash" it's not too helpful - you >need to at least find out where it's crashing, and why. Thank you again Mike. As soon as you said refcount, I realised where my mistake was. First was I was doing a unref of the buffer when it was not required. Second, and the more grave mistake was, I was not setting the CAPS for the buffer. I was under the impression CAPS will be taken from the pads, but I was wrong. Now gst_pad_pull_range works brilliantly for me, without any memcpy or delay. Thanks again. Rahul Verma SASKEN BUSINESS DISCLAIMER ------------------------- This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, Disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email From julien.pauty at gmail.com Tue Jan 27 15:26:06 2009 From: julien.pauty at gmail.com (Julien Pauty) Date: Tue, 27 Jan 2009 15:26:06 +0100 Subject: [gst-devel] Help converting a saved transport stream in an avi file Message-ID: Hello, I would like to demux a transport stream and remux it into an avi file. I'm using this pipeline: gst-launch-0.10 avimux name=mux ! filesink location=/tmp/test2.avi sync=true filesrc location=jeu.mpg do-timestamp=true ! ffdemux_mpegts name=demux demux.video_00 ! queue ! mux.video_00 demux.audio_00 ! queue ! mux.audio_00 This partially works: sound is perfect, but video is slow. It's like if many frames are dropped. I thought that this transformation should just change the container and should not affect frame rate. Is it possible to do this with gstreamer? Thanks, Best regards, Julien -------------- next part -------------- An HTML attachment was scrubbed... URL: From msmith at xiph.org Tue Jan 27 17:11:29 2009 From: msmith at xiph.org (Michael Smith) Date: Tue, 27 Jan 2009 08:11:29 -0800 Subject: [gst-devel] How to register Gstreamer Plugin with Playbin so that playbin detect my plugins In-Reply-To: <21683555.post@talk.nabble.com> References: <21541362.post@talk.nabble.com> <785339900901200222g61183847q4af355dc76c1a65c@mail.gmail.com> <3c1737210901201009s219c7e52rac3a81b8056affbd@mail.gmail.com> <21578039.post@talk.nabble.com> <1232528128.13038.4.camel@mini.centricular.net> <21582459.post@talk.nabble.com> <1232543301.22921.11.camel@mini.centricular.net> <21599063.post@talk.nabble.com> <497A4019.1030205@hora-obscura.de> <21683555.post@talk.nabble.com> Message-ID: <3c1737210901270811p2e8f1015x9bb1fb2c031d2677@mail.gmail.com> On Tue, Jan 27, 2009 at 3:18 AM, rakesh sharma wrote: > > Can anybody tell me what this function do in depth... > gst_element_make_from_uri() > the description is like this : Creates an element for handling the given URI > If i want that it will use my own created plugin is it possible ??? > My plugin dont have the uri property, whether i have to add uri property and > do some more changes ??? You plugin will need to implement the GstUriHandler interface (it's customary, but not required, that it also have a uri property). Mike From sebastian.droege at collabora.co.uk Tue Jan 27 20:34:05 2009 From: sebastian.droege at collabora.co.uk (Sebastian =?ISO-8859-1?Q?Dr=F6ge?=) Date: Tue, 27 Jan 2009 20:34:05 +0100 Subject: [gst-devel] [gst-cvs] ensonic gst-plugins-good: gst-plugins-good/ gst-plugins-good/tests/examples/spectrum/ In-Reply-To: <20080810193519.4EE449E747@gabe.freedesktop.org> References: <20080810193519.4EE449E747@gabe.freedesktop.org> Message-ID: <1233084845.561.48.camel@odin.lan> Am Sonntag, den 10.08.2008, 12:35 -0700 schrieb ensonic at kemper.freedesktop.org: > CVS Root: /cvs/gstreamer > Module: gst-plugins-good > Changes by: ensonic > Date: Sun Aug 10 2008 19:35:19 UTC > > Log message: > * tests/examples/spectrum/demo-audiotest.c: > * tests/examples/spectrum/demo-osssrc.c: > Demo how to draw analyzer results synced to the clock. This is not a good idea as the drawing callback will be called from some random thread. On X11 you can put all the drawing code inside the Gdk lock but on Windows the application will simply die as it's simply not allowed there to do GUI stuff from outside the thread that runs the main loop. I'll add the Gdk locking for now to make it work on X11 at least but maybe this change should be reverted :) (On X11 it will sometimes kill the application with assertion failures inside libX11) -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: Dies ist ein digital signierter Nachrichtenteil URL: From sebastian.droege at collabora.co.uk Tue Jan 27 20:36:50 2009 From: sebastian.droege at collabora.co.uk (Sebastian =?ISO-8859-1?Q?Dr=F6ge?=) Date: Tue, 27 Jan 2009 20:36:50 +0100 Subject: [gst-devel] [gst-cvs] ensonic gst-plugins-good: gst-plugins-good/ gst-plugins-good/tests/examples/spectrum/ In-Reply-To: <1233084845.561.48.camel@odin.lan> References: <20080810193519.4EE449E747@gabe.freedesktop.org> <1233084845.561.48.camel@odin.lan> Message-ID: <1233085010.561.50.camel@odin.lan> Am Dienstag, den 27.01.2009, 20:34 +0100 schrieb Sebastian Dr?ge: > Am Sonntag, den 10.08.2008, 12:35 -0700 schrieb > ensonic at kemper.freedesktop.org: > > CVS Root: /cvs/gstreamer > > Module: gst-plugins-good > > Changes by: ensonic > > Date: Sun Aug 10 2008 19:35:19 UTC > > > > Log message: > > * tests/examples/spectrum/demo-audiotest.c: > > * tests/examples/spectrum/demo-osssrc.c: > > Demo how to draw analyzer results synced to the clock. > > This is not a good idea as the drawing callback will be called > from some random thread. On X11 you can put all the drawing > code inside the Gdk lock but on Windows the application > will simply die as it's simply not allowed there to do GUI > stuff from outside the thread that runs the main loop. > > I'll add the Gdk locking for now to make it work on X11 at > least but maybe this change should be reverted :) > > (On X11 it will sometimes kill the application with assertion > failures inside libX11) Actually taking the Gdk lock (gdk_threads_{enter,leave}()) won't help much here: (demo-audiotest:11717): Gdk-WARNING **: /tmp/buildd/gtk +2.0-2.14.7/gdk/gdkwindow.c:1200: no preceding call to gdk_window_begin_paint_region(), see documentation demo-audiotest: ../../src/xcb_lock.c:33: _XCBUnlockDisplay: Zusicherung ?xcb_get_request_sent(dpy->xcb->connection) == dpy->request? nicht erf?llt. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: Dies ist ein digital signierter Nachrichtenteil URL: From ensonic at hora-obscura.de Tue Jan 27 22:20:10 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Tue, 27 Jan 2009 23:20:10 +0200 Subject: [gst-devel] [gst-cvs] ensonic gst-plugins-good: gst-plugins-good/ gst-plugins-good/tests/examples/spectrum/ In-Reply-To: <1233084845.561.48.camel@odin.lan> References: <20080810193519.4EE449E747@gabe.freedesktop.org> <1233084845.561.48.camel@odin.lan> Message-ID: <497F7A8A.6050307@hora-obscura.de> Sebastian Dr?ge schrieb: > Am Sonntag, den 10.08.2008, 12:35 -0700 schrieb > ensonic at kemper.freedesktop.org: >> CVS Root: /cvs/gstreamer >> Module: gst-plugins-good >> Changes by: ensonic >> Date: Sun Aug 10 2008 19:35:19 UTC >> >> Log message: >> * tests/examples/spectrum/demo-audiotest.c: >> * tests/examples/spectrum/demo-osssrc.c: >> Demo how to draw analyzer results synced to the clock. > > This is not a good idea as the drawing callback will be called > from some random thread. On X11 you can put all the drawing > code inside the Gdk lock but on Windows the application > will simply die as it's simply not allowed there to do GUI > stuff from outside the thread that runs the main loop. > > I'll add the Gdk locking for now to make it work on X11 at > least but maybe this change should be reverted :) > > (On X11 it will sometimes kill the application with assertion > failures inside libX11) I actually recently fixed this in buzztard and now also fixed this in the examples. Thanks for pointing out. I also added a comment to the API docs. Honestly speaking we should come up with a saner API for this. - a main loop source that syncs message delivery with the clock? - just a utillity function for delaying the message, but preserving the context? Stefan From donari75 at gmail.com Wed Jan 28 03:23:15 2009 From: donari75 at gmail.com (unsik Kim) Date: Wed, 28 Jan 2009 11:23:15 +0900 Subject: [gst-devel] [gst-plugins-base-0.10.20] alsasink can't get appropiate sound format Message-ID: <57afda040901271823r1245da82h9b7d1a2f7fd769a@mail.gmail.com> Hello, all? I have trouble with alsasink plugin (I just guessed, I'm a newbie). First, my circumstances are, gst-plugins-base-0.10.20 gstreamer-0.10.20 alsa-lib-1.0.15 kernel 2.6.29-rc2 CPU : PXA270 H/W audio codec : UCB1400 AC97 codec My problem is alsasink dosen't produce appropriate sound. Left channel : noise (simillar with white noise) Right channel : play sound but slow (2 times slower... I guess) First I checked operation of sound device with speaker-test of alsa-utils and it works well. Here is log. --- root at mgpmp:~# speaker-test -c 2 -t wav speaker-test 1.0.15 Playback device is default Stream parameters are 48000Hz, S16_LE, 2 channels WAV file(s) Rate set to 48000Hz (requested 48000Hz) Buffer size range from 8 to 32768 Period size range from 8 to 2040 Using max buffer size 32768 Periods = 4 was set period_size = 1024 was set buffer_size = 32768 0 - Front Left 1 - Front Right --- Also, I can play well with ALSA's oss compatibility module (snd-oss-pcm). Here is gstreamers log. (long) --- root at mgpmp:~# gst-launch playbin uri=file:///home/root/07\ Red\ Dress.mp3 Setting pipeline to PAUSED ... 0:00:00.933727077 716 0x15050 DEBUG alsa gstalsaplugin.c:75:plugin_init: binding text domain gst-plugins-base-0.10 to locale dir /usr/share/locale 0:00:00.940897231 716 0x15050 DEBUG alsa gstalsasink.c:276:gst_alsasink_init: initializing alsasink 0:00:00.942173538 716 0x15050 DEBUG alsa gstalsasink.c:306:gst_alsasink_getcaps: device not open, using template caps 0:00:01.001892308 716 0x15050 LOG alsa gstalsasink.c:681:gst_alsasink_open: Opened device default 0:00:01.059183077 716 0x15050 WARN alsa gstalsa.c:124:gst_alsa_detect_formats: skipping non-int format 0:00:01.060123077 716 0x15050 LOG alsa gstalsa.c:30:gst_alsa_detect_rates: probing sample rates ... 0:00:01.060633538 716 0x15050 DEBUG alsa gstalsa.c:49:gst_alsa_detect_rates: Min. rate = 4000 (4000) 0:00:01.061093538 716 0x15050 DEBUG alsa gstalsa.c:50:gst_alsa_detect_rates: Max. rate = 2147483647 (-1) 0:00:01.061746461 716 0x15050 LOG alsa gstalsa.c:265:gst_alsa_detect_channels: probing channels ... 0:00:01.062253846 716 0x15050 DEBUG alsa gstalsa.c:309:gst_alsa_detect_channels: Min. channels = 1 (1) 0:00:01.062769538 716 0x15050 DEBUG alsa gstalsa.c:310:gst_alsa_detect_channels: Max. channels = 8 (10000) 0:00:01.069656308 716 0x15050 DEBUG alsa gstalsa.c:388:gst_alsa_open_iec958_pcm: Generated device string "iec958:{AES0 0x02 AES1 0x82 AES2 0x00 AES3 0x02}" 0:00:01.070788308 716 0x15050 WARN alsa conf.c:3949:snd_config_expand: alsalib error: Unknown parameters {AES0 0x02 AES1 0x82 AES2 0x00 AES3 0x02} 0:00:01.071308000 716 0x15050 WARN alsa pcm.c:2145:snd_pcm_open_noupdate: alsalib error: Unknown PCM iec958:{AES0 0x02 AES1 0x82 AES2 0x00 AES3 0x02} 0:00:01.071997231 716 0x15050 DEBUG alsa gstalsa.c:394:gst_alsa_open_iec958_pcm: failed opening IEC958 device: Invalid argument 0:00:01.072552615 716 0x15050 INFO alsa gstalsasink.c:326:gst_alsasink_getcaps: returning caps audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)32, depth=(int)32, rate=(int)[ 4000, 2147483647 ], channels=(int)[ 1, 2 ]; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)32, depth=(int)32, rate=(int)[ 4000, 2147483647 ], channels=(int)3, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)32, depth=(int)32, rate=(int)[ 4000, 2147483647 ], channels=(int)4, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT >; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)32, depth=(int)32, rate=(int)[ 4000, 2147483647 ], channels=(int)6, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)32, depth=(int)32, rate=(int)[ 4000, 2147483647 ], channels=(int)8, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT >; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)24, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)[ 1, 2 ]; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)24, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)3, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)24, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)4, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT >; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)24, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)6, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)24, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)8, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT >; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)32, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)[ 1, 2 ]; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)32, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)3, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)32, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)4, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT >; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)32, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)6, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)32, depth=(int)24, rate=(int)[ 4000, 2147483647 ], channels=(int)8, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT >; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)16, depth=(int)16, rate=(int)[ 4000, 2147483647 ], channels=(int)[ 1, 2 ]; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)16, depth=(int)16, rate=(int)[ 4000, 2147483647 ], channels=(int)3, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)16, depth=(int)16, rate=(int)[ 4000, 2147483647 ], channels=(int)4, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT >; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)16, depth=(int)16, rate=(int)[ 4000, 2147483647 ], channels=(int)6, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int)16, depth=(int)16, rate=(int)[ 4000, 2147483647 ], channels=(int)8, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT >; audio/x-raw-int, signed=(boolean){ true, false }, width=(int)8, depth=(int)8, rate=(int)[ 4000, 2147483647 ], channels=(int)[ 1, 2 ]; audio/x-raw-int, signed=(boolean){ true, false }, width=(int)8, depth=(int)8, rate=(int)[ 4000, 2147483647 ], channels=(int)3, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, signed=(boolean){ true, false }, width=(int)8, depth=(int)8, rate=(int)[ 4000, 2147483647 ], channels=(int)4, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT >; audio/x-raw-int, signed=(boolean){ true, false }, width=(int)8, depth=(int)8, rate=(int)[ 4000, 2147483647 ], channels=(int)6, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE >; audio/x-raw-int, signed=(boolean){ true, false }, width=(int)8, depth=(int)8, rate=(int)[ 4000, 2147483647 ], channels=(int)8, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT > 0:00:01.780125846 716 0x15050 LOG alsa gstalsasink.c:311:gst_alsasink_getcaps: Returning cached caps Pipeline is PREROLLING ... 0:00:02.020555692 716 0x62888 LOG alsa gstalsasink.c:311:gst_alsasink_getcaps: Returning cached caps 0:00:02.121833231 716 0x62888 LOG alsa gstalsasink.c:311:gst_alsasink_getcaps: Returning cached caps 0:00:02.265676000 716 0x62888 LOG alsa gstalsasink.c:311:gst_alsasink_getcaps: Returning cached caps 0:00:02.403471693 716 0xba968 LOG alsa gstalsasink.c:311:gst_alsasink_getcaps: Returning cached caps 0:00:02.482740615 716 0x62888 LOG alsa gstalsasink.c:311:gst_alsasink_getcaps: Returning cached caps 0:00:02.677360308 716 0xba968 LOG alsa gstalsasink.c:311:gst_alsasink_getcaps: Returning cached caps 0:00:02.741654461 716 0x62888 LOG alsa gstalsasink.c:311:gst_alsasink_getcaps: Returning cached caps 0:00:02.858667077 716 0xba968 LOG alsa gstalsasink.c:311:gst_alsasink_getcaps: Returning cached caps 0:00:02.982380000 716 0xba968 LOG alsa gstalsasink.c:311:gst_alsasink_getcaps: Returning cached caps 0:00:03.110284923 716 0xba968 LOG alsa gstalsasink.c:311:gst_alsasink_getcaps: Returning cached caps 0:00:03.218500308 716 0xba968 LOG alsa gstalsasink.c:311:gst_alsasink_getcaps: Returning cached caps 0:00:03.262078461 716 0xba968 LOG alsa gstalsasink.c:311:gst_alsasink_getcaps: Returning cached caps 0:00:03.324798769 716 0xba968 DEBUG alsa gstalsasink.c:619:alsasink_parse_spec: Linear format : depth=32, width=32, sign=1, bigend=0 0:00:03.325648308 716 0xba968 DEBUG alsa gstalsasink.c:343:set_hwparams: Negotiating to 2 channels @ 44100 Hz (format = S32_LE) SPDIF (0) 0:00:03.329065231 716 0xba968 DEBUG alsa gstalsasink.c:387:set_hwparams: buffer time 200000, min 90, max 371520 0:00:03.329612923 716 0xba968 DEBUG alsa gstalsasink.c:393:set_hwparams: period time 10000, min 90, max 23130 0:00:03.330100615 716 0xba968 DEBUG alsa gstalsasink.c:398:set_hwparams: periods min 0, max 4096 0:00:03.331443077 716 0xba968 DEBUG alsa gstalsasink.c:414:set_hwparams: buffer time 200000 0:00:03.333042770 716 0xba968 DEBUG alsa gstalsasink.c:427:set_hwparams: period time 9523 0:00:03.353240308 716 0xba968 DEBUG alsa gstalsasink.c:451:set_hwparams: buffer size 8820, period size 420 0:00:03.354189539 716 0xba968 DEBUG alsa gstalsasink.c:738:gst_alsasink_prepare: Hardware setup: stream : PLAYBACK access : RW_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 32 buffer_size : 8820 period_size : 420 period_time : 9523 tick_time : 0 0:00:03.354938154 716 0xba968 DEBUG alsa gstalsasink.c:743:gst_alsasink_prepare: Software setup: tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 420 xfer_align : 1 start_threshold : 8820 stop_threshold : 8820 silence_threshold: 0 silence_size : 0 boundary : 1156055040 Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstAudioSinkClock 0:00:03.430652923 716 0x5f658 LOG alsa gstalsasink.c:893:gst_alsasink_write: received audio samples buffer of 3360 bytes 0:00:03.431441231 716 0x5f658 DEBUG alsa gstalsasink.c:901:gst_alsasink_write: written 420 frames out of 420 --- Currently, I think " format : S32_LE" is wrong. " format : S16_LE" will be adequate. But I can't determine what is the cause of fault (alsasink, somewhat outdated UCB1400 driver, ALSA-lib). I want to someone advise me for proper debugging points. Any little help will be really appreciated. unsik Kim From sumanth.v at allaboutif.com Wed Jan 28 09:12:02 2009 From: sumanth.v at allaboutif.com (Sumanth V) Date: Wed, 28 Jan 2009 13:42:02 +0530 Subject: [gst-devel] DVD playback using rsndvdbin Message-ID: <83b2c1480901280012w1455ba1dw2133f4dc3bd768ec@mail.gmail.com> Hi all, I am writing a application for DVD playback using the rsndvdbin. I edited the resindvdbin.c code so that it can handle different audio format. i am not using the decodebin. I tried playing the DVD's having different audio format and it worked fine. But when i tried playing a DVD in which one list of Video songs had a audio format( audio/x-private1-ac3) and the other List of songs were in "audio/mpeg" format, the 'dvddemux' dose not generate the pad-added signal correctly. The 'dvddemux' generate the pad-added signal only once for different audio format. i.e:- If its playing a video with audio format of "audio/x-private1-ac3" and now if i select to play a video with audio format "audio/mpeg" dvddemux generate the 'pad-added' signal. But now if i want to play a video which has a audio format of "audio/x-private1-ac3" the dvddemux dose not generate the pad-added signal. Because of which i am not able to play the audio, but the video is been played. But if i try playing the video with "audio/mpeg" audio format it works fine, since the pad is allreay linked. Do i have to make any changes in the "gstmpegdemux.c" and "resindvdsrc.c" ?? Thanks all. -------------- next part -------------- An HTML attachment was scrubbed... URL: From sabine.donze at gmail.com Wed Jan 28 15:12:02 2009 From: sabine.donze at gmail.com (=?ISO-8859-1?Q?sabine_donz=E9?=) Date: Wed, 28 Jan 2009 15:12:02 +0100 Subject: [gst-devel] equalizer audio Message-ID: Hello, I'm using the equalizer-nbands plugin from gst-plugins-good_0.10.7. However, if I configure the frequency for one or for each band, the observed cutoff frequency is not the same than the defined frequency. For example: gst-launch filesrc location=toto.mp3 ! ffdemux_mp3 ! mfw_mp3decoder ! equalizer-nbands num-bands=4 band0::gain=-21 band0::frequency=947 band0::bandwidth=100 ! alsasink I should have a cutting frequency at 957Hz for band0. In logfile, I observe that gain, frequency and bandwidth have been correctly changed. But in setup_filter function, frequency is not at defined value: 0:00:00.516575000 1477 0x14018 DEBUG equalizer gstiirequalizer.c:494:gst_iir_equalizer_compute_frequencies: band[ 0] = '66.234133' 0:00:00.517031000 1477 0x14018 DEBUG equalizer gstiirequalizer.c:494:gst_iir_equalizer_compute_frequencies: band[ 1] = '372.461899' 0:00:00.517357000 1477 0x14018 DEBUG equalizer gstiirequalizer.c:494:gst_iir_equalizer_compute_frequencies: band[ 2] = '2094.507176' 0:00:00.517678000 1477 0x14018 DEBUG equalizer gstiirequalizer.c:494:gst_iir_equalizer_compute_frequencies: band[ 3] = '11778.279410' 0:00:00.518174000 1477 0x14018 LOG equalizer gstiirequalizer.c:290:gst_iir_equalizer_child_proxy_get_children_count: we have 4 children 0:00:00.518600000 1477 0x14018 LOG equalizer gstiirequalizer.c:281:gst_iir_equalizer_child_proxy_get_child_by_index: return child[0] 'band0' 0:00:00.519124000 1477 0x14018 DEBUG equalizer gstiirequalizer.c:136:gst_iir_equalizer_band_set_property: gain = 0.000000 -> -21.000000 0:00:00.519571000 1477 0x14018 DEBUG equalizer gstiirequalizer.c:146:gst_iir_equalizer_band_set_property: changed gain = -21.000000 0:00:00.520126000 1477 0x14018 LOG equalizer gstiirequalizer.c:290:gst_iir_equalizer_child_proxy_get_children_count: we have 4 children 0:00:00.520444000 1477 0x14018 LOG equalizer gstiirequalizer.c:281:gst_iir_equalizer_child_proxy_get_child_by_index: return child[0] 'band0' 0:00:00.520852000 1477 0x14018 DEBUG equalizer gstiirequalizer.c:154:gst_iir_equalizer_band_set_property: freq = 66.234133 -> 947.000000 0:00:00.521334000 1477 0x14018 DEBUG equalizer gstiirequalizer.c:164:gst_iir_equalizer_band_set_property: changed freq = 947.000000 0:00:00.521767000 1477 0x14018 LOG equalizer gstiirequalizer.c:290:gst_iir_equalizer_child_proxy_get_children_count: we have 4 children 0:00:00.522071000 1477 0x14018 LOG equalizer gstiirequalizer.c:281:gst_iir_equalizer_child_proxy_get_child_by_index: return child[0] 'band0' 0:00:00.522568000 1477 0x14018 DEBUG equalizer gstiirequalizer.c:172:gst_iir_equalizer_band_set_property: width = 92.468265 -> 100.000000 0:00:00.523032000 1477 0x14018 DEBUG equalizer gstiirequalizer.c:182:gst_iir_equalizer_band_set_property: changed width = 100.000000 Setting pipeline to PAUSED ... Pipeline is PREROLLING ... 0:00:00.996510000 1477 0x2e7d0 DEBUG equalizer gstiirequalizer.c:451:gst_iir_equalizer_compute_frequencies: bands 4 -> 4 0:00:00.997166000 1477 0x2e7d0 DEBUG equalizer gstiirequalizer.c:494:gst_iir_equalizer_compute_frequencies: band[ 0] = '66.234133' 0:00:00.997577000 1477 0x2e7d0 DEBUG equalizer gstiirequalizer.c:494:gst_iir_equalizer_compute_frequencies: band[ 1] = '372.461899' 0:00:00.997921000 1477 0x2e7d0 DEBUG equalizer gstiirequalizer.c:494:gst_iir_equalizer_compute_frequencies: band[ 2] = '2094.507176' 0:00:00.998256000 1477 0x2e7d0 DEBUG equalizer gstiirequalizer.c:494:gst_iir_equalizer_compute_frequencies: band[ 3] = '11778.279410' 0:00:00.998887000 1477 0x2e7d0 INFO equalizer gstiirequalizer.c:438:setup_filter: gain = -21, , bandwidth= 92.468, frequency = 66.234, a0 = 0.98034, 0:00:00.999431000 1477 0x2e7d0 INFO equalizer gstiirequalizer.c:438:setup_filter: gain = 0, , bandwidth= 519.99, frequency = 372.46, a0 = 1, a1 0:00:01.000021000 1477 0x2e7d0 INFO equalizer gstiirequalizer.c:438:setup_filter: gain = 0, , bandwidth= 2924.1, frequency = 2094.5, a0 = 1, a1 0:00:01.000526000 1477 0x2e7d0 INFO equalizer gstiirequalizer.c:438:setup_filter: gain = 0, , bandwidth= 16443, frequency = 11778, a0 = 1, a1 Moreover, cutoff frequency observed at the end of audio chain is around 200Hz I've also tried to define all bands, but cutting frequencies observed are not coherent with defined frequencies. Moreover, according to configurations, we can have a difference of 6dB between right way and left way. How does this plugin work? How configure it so as to have a cutting frequency at 947Hz? How to have the same output for right way and left way, or how to configure the difference between right way and left way? Thanks for your help Sabine -------------- next part -------------- An HTML attachment was scrubbed... URL: From arnout at mind.be Wed Jan 28 16:25:43 2009 From: arnout at mind.be (Arnout Vandecappelle) Date: Wed, 28 Jan 2009 16:25:43 +0100 Subject: [gst-devel] Saving a live stream In-Reply-To: <497AD842.8040706@heydan.co.uk> References: <497AD842.8040706@heydan.co.uk> Message-ID: <200901281625.43599.arnout@mind.be> On Saturday 24 January 2009 09:58:42 Hayden Andrews wrote: > First of all, I wrote a quick Perl script which reads directly from the > appropriate HTTP port (an Icecast Server) and writes to a file. > Everything looks fine with this file except .. > 1) Suppose we record from 40 mins to 1:20 of the live stream. When we > play the file, the current time starts at the time we started recording > from rather than from 00:00. Although the run time is correct (40 mins), > the current time progresses from 40mins to 1:20. That's because the buffer timestamps are retained. You have to re-stamp the buffers. I ended up writing an element to do exactly that: http://bugzilla.gnome.org/show_bug.cgi?id=561224 - that bugzilla entry also contains a remark on how it should be done by setting base_time, but I don't think that will work. As discussed earlier on the list (see http://thread.gmane.org/gmane.comp.video.gstreamer.devel/22806 ), you can also try an identity with single-segment=true, although I also doubt that will work :-). In any case, you'll need to do demuxing and remuxing to get the timestamps in your output file. So your pipeline will look like this: filesrc location=... ! oggdemux ! identity single-segment=true \ ! oggmux ! filesink location=... Of course, you'll encounter additional problems with audio: you'll need to split the pipeline and add queue's, and I'm pretty sure something will go wrong with the synchronisation between the timestamps of the audio and video channel. > 2) When someone tries to edit the file to trim each end and make any > other minor changes that are required, seeking takes forever and the > editor takes for ever to do things that would normally be quite quick > with a normal file. I believe that if the file has a play time of about > 40 mins, then the player says that it will take about 40 mins to do many > tasks. > > I figure that since a live stream has an unknown length, and since it is > not possible to seek when viewing a live stream, that there is missing > info in the live stream that needs to be inserted while saving. So, I > also figure that it would be easy to insert the relevant requirements > while saving a live stream to file..... Indeed, the output file doesn't contain an index, which means the editor needs to scan the entire file in order to seek correctly. The index is only written when the muxer receives an EOS event. Unfortunately, as far as I know, there is no way to force an EOS on a stream started with gst-launch. How do you select the part you want using GStreamer anyway? It may pay off to take a look at gentrans: http://gentrans.sourceforge.net/ Regards, Arnout -- Arnout Vandecappelle arnout at mind be Senior Embedded Software Architect +32-16-286540 Essensium/Mind http://www.mind.be G.Geenslaan 9, 3001 Leuven, Belgium BE 872 984 063 RPR Leuven LinkedIn profile: http://www.linkedin.com/in/arnoutvandecappelle GPG fingerprint: D206 D44B 5155 DF98 550D 3F2A 2213 88AA A1C7 C933 From dstaples at friendmts.co.uk Wed Jan 28 17:00:32 2009 From: dstaples at friendmts.co.uk (Darren Staples) Date: Wed, 28 Jan 2009 16:00:32 +0000 Subject: [gst-devel] Appsrc question Message-ID: Can you tell me please, in a video processing pipeline, is the appsrc component used to feed video frames into the pipeline, or raw video stream data ? Thanks ! Darren *** FRIEND MTS LIMITED EMAIL CONFIDENTIALITY *** This email message and any attachments may contain information which is confidential or privileged and is intended for the sole use of the person to whom it is addressed. If you are not the intended recipient, be aware that any disclosures, copying, distribution or use of the contents is prohibited. If you have received this email message in error, please notify our office by telephone (+44 (0)121 633 2000) or email (admin at friendmts.co.uk) immediately. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: From thaytan at noraisin.net Wed Jan 28 17:36:02 2009 From: thaytan at noraisin.net (Jan Schmidt) Date: Wed, 28 Jan 2009 16:36:02 +0000 Subject: [gst-devel] DVD playback using rsndvdbin In-Reply-To: <83b2c1480901280012w1455ba1dw2133f4dc3bd768ec@mail.gmail.com> References: <83b2c1480901280012w1455ba1dw2133f4dc3bd768ec@mail.gmail.com> Message-ID: <1233160562.31252.1.camel@fancy-ubuntu> On Wed, 2009-01-28 at 13:42 +0530, Sumanth V wrote: > Hi all, > > I am writing a application for DVD playback using the > rsndvdbin. I edited the resindvdbin.c code so that it can handle > different audio format. i am not using the decodebin. I tried playing > the DVD's having different audio format and it worked fine. But when i > tried playing a DVD in which one list of Video songs had a audio > format( audio/x-private1-ac3) and the other List of songs were in > "audio/mpeg" format, the 'dvddemux' dose not generate the pad-added > signal correctly. *snip* > Do i have to make any changes in the "gstmpegdemux.c" and > "resindvdsrc.c" ?? There's a list of the current development status, and all the things I haven't yet done for DVD playback at http://gstreamer.freedesktop.org/wiki/DvdPlayback The feature you're looking at comes under the 'Switching Audio Decoder Bin' heading. J. -- Jan Schmidt From thiagossantos at gmail.com Wed Jan 28 17:47:32 2009 From: thiagossantos at gmail.com (thiagoss) Date: Wed, 28 Jan 2009 14:47:32 -0200 Subject: [gst-devel] Building failing for qtwrapper In-Reply-To: <497B3E34.6050604@hora-obscura.de> References: <497B3E34.6050604@hora-obscura.de> Message-ID: On Sat, Jan 24, 2009 at 2:13 PM, Stefan Kost wrote: > thiagoss schrieb: > > Qtwraper building is failing (cvs head), I don't know if this is a > > misconfiguration in my system or in Makefile, so I'm sending this mail. > > The error output is: > > > > Making all in qtwrapper > > cd ../.. && /bin/sh > > /Users/thiagoss/Work/GStreamer/head/gst-plugins-bad/missing --run > > automake-1.10 --gnu sys/qtwrapper/Makefile > > configure.ac:111 : warning: > > AC_CACHE_VAL(g_support_dowhile_macros, ...): suspicious cache-id, must > > contain _cv_ to be cached > > ../../lib/autoconf/general.m4:1974: AC_CACHE_VAL is expanded from... > > ../../lib/autoconf/general.m4:1994: AC_CACHE_CHECK is expanded from... > > common/m4/gst-dowhile.m4:8: AG_GST_CHECK_DOWHILE_MACROS is expanded > from... > > configure.ac:111 : the top level > > configure.ac:143 : warning: > > AC_CACHE_VAL(GST_FIONREAD_IN_SYS_IOCTL, ...): suspicious cache-id, must > > contain _cv_ to be cached > > m4/gst-fionread.m4:1: GST_CHECK_FIONREAD is expanded from... > > configure.ac:143 : the top level > > configure.ac:143 : warning: > > AC_CACHE_VAL(GST_FIONREAD_IN_SYS_FILIO, ...): suspicious cache-id, must > > contain _cv_ to be cached > > cd ../.. && /bin/sh ./config.status sys/qtwrapper/Makefile depfiles > > config.status: creating sys/qtwrapper/Makefile > > config.status: executing depfiles commands > > /bin/sh ../../libtool --tag=CC --tag=disable-static --mode=compile gcc > > -DHAVE_CONFIG_H -I. -I../.. -I../../gst-libs -I../../gst-libs > > -D_REENTRANT -I/usr/local/include/gstreamer-0.10 > > -I/opt/local/include/glib-2.0 -I/opt/local/lib/glib-2.0/include > > -I/opt/local/include -I/opt/local/include/libxml2 -Wall > > -Wdeclaration-after-statement -Werror -fno-strict-aliasing -g > > -D_REENTRANT -I/usr/local/include/gstreamer-0.10 > > -I/opt/local/include/glib-2.0 -I/opt/local/lib/glib-2.0/include > > -I/opt/local/include -I/opt/local/include/libxml2 -g -O2 -MT > > libgstqtwrapper_la-qtwrapper.lo -MD -MP -MF > > .deps/libgstqtwrapper_la-qtwrapper.Tpo -c -o > > libgstqtwrapper_la-qtwrapper.lo `test -f 'qtwrapper.c' || echo > > './'`qtwrapper.c > > libtool: compile: gcc -DHAVE_CONFIG_H -I. -I../.. -I../../gst-libs > > -I../../gst-libs -D_REENTRANT -I/usr/local/include/gstreamer-0.10 > > -I/opt/local/include/glib-2.0 -I/opt/local/lib/glib-2.0/include > > -I/opt/local/include -I/opt/local/include/libxml2 -Wall > > -Wdeclaration-after-statement -Werror -fno-strict-aliasing -g > > -D_REENTRANT -I/usr/local/include/gstreamer-0.10 > > -I/opt/local/include/glib-2.0 -I/opt/local/lib/glib-2.0/include > > -I/opt/local/include -I/opt/local/include/libxml2 -g -O2 -MT > > libgstqtwrapper_la-qtwrapper.lo -MD -MP -MF > > .deps/libgstqtwrapper_la-qtwrapper.Tpo -c qtwrapper.c -fno-common -DPIC > > -o .libs/libgstqtwrapper_la-qtwrapper.o > > In file included from qtwrapper.c:49: > > qtwrapper.h:53:20: error: Movies.h: No such file or directory > > qtwrapper.c: In function 'plugin_init': > > qtwrapper.c:58: error: 'OSErr' undeclared (first use in this function) > > qtwrapper.c:58: error: (Each undeclared identifier is reported only once > > qtwrapper.c:58: error: for each function it appears in.) > > qtwrapper.c:58: error: syntax error before 'status' > > qtwrapper.c:69: error: 'status' undeclared (first use in this function) > > cc1: warnings being treated as errors > > qtwrapper.c:69: warning: implicit declaration of function 'EnterMovies' > > make[3]: *** [libgstqtwrapper_la-qtwrapper.lo] Error 1 > > make[2]: *** [all-recursive] Error 1 > > make[1]: *** [all-recursive] Error 1 > > make: *** [all] Error 2 > > > > How to fix this? > > It seems that configure belives you have all the needed external libs and > headers, but in fact you don't. Do you have a Movies.h? Maybe you can look > at > config.log and check what its looking for. > > Stefan > # find / -name "Movies.h" /Developer/Headers/FlatCarbon/Movies.h /Developer/SDKs/MacOSX10.4u.sdk/Developer/Headers/CFMCarbon/Movies.h /Developer/SDKs/MacOSX10.4u.sdk/Developer/Headers/CFMCarbon/QuickTime/Movies.h /Developer/SDKs/MacOSX10.4u.sdk/Developer/Headers/FlatCarbon/Movies.h /Developer/SDKs/MacOSX10.4u.sdk/System/Library/Frameworks/QuickTime.framework/Versions/A/Headers/Movies.h /Developer/SDKs/MacOSX10.5.sdk/Developer/Headers/FlatCarbon/Movies.h /Developer/SDKs/MacOSX10.5.sdk/System/Library/Frameworks/QuickTime.framework/Versions/A/Headers/Movies.h /System/Library/Frameworks/QuickTime.framework/Versions/A/Headers/Movies.h Maybe it is looking in the wrong places. The config.log has this: configure:30272: *** checking feature: QuickTime wrapper *** configure:30276: *** for plug-ins: qtwrapper *** configure:30324: checking QuickTime/Movies.h usability configure:30341: gcc -c -g -O2 conftest.c >&5 configure:30348: $? = 0 configure:30362: result: yes configure:30366: checking QuickTime/Movies.h presence configure:30381: gcc -E conftest.c configure:30388: $? = 0 configure:30402: result: yes configure:30435: checking for QuickTime/Movies.h configure:30442: result: yes configure:30464: *** These plugins will be built: qtwrapper > > > > > -- > > Thiago Sousa Santos > > > > Embedded Systems and Pervasive Computing Lab (Embedded) > > Center of Electrical Engineering and Informatics (CEEI) > > Federal University of Campina Grande (UFCG) > > > > > > ------------------------------------------------------------------------ > > > > > ------------------------------------------------------------------------------ > > This SF.net email is sponsored by: > > SourcForge Community > > SourceForge wants to tell your story. > > http://p.sf.net/sfu/sf-spreadtheword > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Thiago Sousa Santos Embedded Systems and Pervasive Computing Lab (Embedded) Center of Electrical Engineering and Informatics (CEEI) Federal University of Campina Grande (UFCG) -------------- next part -------------- An HTML attachment was scrubbed... URL: From saicharmboy at yahoo.co.in Wed Jan 28 18:00:30 2009 From: saicharmboy at yahoo.co.in (sai charmboy) Date: Wed, 28 Jan 2009 22:30:30 +0530 (IST) Subject: [gst-devel] (no subject) References: <497B3E34.6050604@hora-obscura.de> Message-ID: <595045.24291.qm@web94814.mail.in2.yahoo.com> shut up all of u.....plz saicharmboy... ________________________________ From: thiagoss To: Discussion of the development of GStreamer Sent: Wednesday, 28 January, 2009 10:17:32 PM Subject: Re: [gst-devel] Building failing for qtwrapper On Sat, Jan 24, 2009 at 2:13 PM, Stefan Kost wrote: thiagoss schrieb: > Qtwraper building is failing (cvs head), I don't know if this is a > misconfiguration in my system or in Makefile, so I'm sending this mail.. > The error output is: > > Making all in qtwrapper > cd ../.. && /bin/sh > /Users/thiagoss/Work/GStreamer/head/gst-plugins-bad/missing --run > automake-1.10 --gnu sys/qtwrapper/Makefile > configure.ac:111 : warning: > AC_CACHE_VAL(g_support_dowhile_macros, ...): suspicious cache-id, must > contain _cv_ to be cached > ../.../lib/autoconf/general.m4:1974: AC_CACHE_VAL is expanded from... > ../../lib/autoconf/general.m4:1994: AC_CACHE_CHECK is expanded from... > common/m4/gst-dowhile.m4:8: AG_GST_CHECK_DOWHILE_MACROS is expanded from... > configure.ac:111 : the top level > configure.ac:143 : warning: > AC_CACHE_VAL(GST_FIONREAD_IN_SYS_IOCTL, ...): suspicious cache-id, must > contain _cv_ to be cached > m4/gst-fionread.m4:1: GST_CHECK_FIONREAD is expanded from... > configure.ac:143 : the top level > configure.ac:143 : warning: > AC_CACHE_VAL(GST_FIONREAD_IN_SYS_FILIO, ...): suspicious cache-id, must > contain _cv_ to be cached > cd .../.. && /bin/sh ./config.status sys/qtwrapper/Makefile depfiles > config.status: creating sys/qtwrapper/Makefile > config.status: executing depfiles commands > /bin/sh ../../libtool --tag=CC --tag=disable-static --mode=compile gcc > -DHAVE_CONFIG_H -I. -I../.. -I../../gst-libs -I../../gst-libs > -D_REENTRANT -I/usr/local/include/gstreamer-0.10 > -I/opt/local/include/glib-2.0 -I/opt/local/lib/glib-2.0/include > -I/opt/local/include -I/opt/local/include/libxml2 -Wall > -Wdeclaration-after-statement -Werror -fno-strict-aliasing -g > -D_REENTRANT -I/usr/local/include/gstreamer-0.10 > -I/opt/local/include/glib-2.0 -I/opt/local/lib/glib-2.0/include > -I/opt/local/include -I/opt/local/include/libxml2 -g -O2 -MT > libgstqtwrapper_la-qtwrapper.lo -MD -MP -MF > .deps/libgstqtwrapper_la-qtwrapper.Tpo -c -o > libgstqtwrapper_la-qtwrapper.lo `test -f 'qtwrapper.c' || echo > './'`qtwrapper.c > libtool: compile: gcc -DHAVE_CONFIG_H -I. -I../.. -I../../gst-libs > -I../../gst-libs -D_REENTRANT -I/usr/local/include/gstreamer-0.10 > -I/opt/local/include/glib-2.0 -I/opt/local/lib/glib-2.0/include > -I/opt/local/include -I/opt/local/include/libxml2 -Wall > -Wdeclaration-after-statement -Werror -fno-strict-aliasing -g > -D_REENTRANT -I/usr/local/include/gstreamer-0.10 > -I/opt/local/include/glib-2.0 -I/opt/local/lib/glib-2.0/include > -I/opt/local/include -I/opt/local/include/libxml2 -g -O2 -MT > libgstqtwrapper_la-qtwrapper.lo -MD -MP -MF > .deps/libgstqtwrapper_la-qtwrapper.Tpo -c qtwrapper.c -fno-common -DPIC > -o .libs/libgstqtwrapper_la-qtwrapper.o > In file included from qtwrapper.c:49: > qtwrapper.h:53:20: error: Movies.h: No such file or directory > qtwrapper.c: In function 'plugin_init': > qtwrapper.c:58: error: 'OSErr' undeclared (first use in this function) > qtwrapper.c:58: error: (Each undeclared identifier is reported only once > qtwrapper.c:58: error: for each function it appears in.) > qtwrapper.c:58: error: syntax error before 'status' > qtwrapper.c:69: error: 'status' undeclared (first use in this function) > cc1: warnings being treated as errors > qtwrapper.c:69: warning: implicit declaration of function 'EnterMovies' > make[3]: *** [libgstqtwrapper_la-qtwrapper.lo] Error 1 > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > How to fix this? It seems that configure belives you have all the needed external libs and headers, but in fact you don't. Do you have a Movies.h? Maybe you can look at config.log and check what its looking for. Stefan # find / -name "Movies.h" /Developer/Headers/FlatCarbon/Movies.h /Developer/SDKs/MacOSX10.4u.sdk/Developer/Headers/CFMCarbon/Movies.h /Developer/SDKs/MacOSX10.4u.sdk/Developer/Headers/CFMCarbon/QuickTime/Movies.h /Developer/SDKs/MacOSX10.4u.sdk/Developer/Headers/FlatCarbon/Movies.h /Developer/SDKs/MacOSX10.4u.sdk/System/Library/Frameworks/QuickTime.framework/Versions/A/Headers/Movies.h /Developer/SDKs/MacOSX10.5.sdk/Developer/Headers/FlatCarbon/Movies.h /Developer/SDKs/MacOSX10.5.sdk/System/Library/Frameworks/QuickTime.framework/Versions/A/Headers/Movies.h /System/Library/Frameworks/QuickTime.framework/Versions/A/Headers/Movies.h Maybe it is looking in the wrong places. The config.log has this: configure:30272: *** checking feature: QuickTime wrapper *** configure:30276: *** for plug-ins: qtwrapper *** configure:30324: checking QuickTime/Movies.h usability configure:30341: gcc -c -g -O2 conftest.c >&5 configure:30348: $? = 0 configure:30362: result: yes configure:30366: checking QuickTime/Movies.h presence configure:30381: gcc -E conftest.c configure:30388: $? = 0 configure:30402: result: yes configure:30435: checking for QuickTime/Movies.h configure:30442: result: yes configure:30464: *** These plugins will be built: qtwrapper > > -- > Thiago Sousa Santos > > Embedded Systems and Pervasive Computing Lab (Embedded) > Center of Electrical Engineering and Informatics (CEEI) > Federal University of Campina Grande (UFCG) > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > > > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel ------------------------------------------------------------------------------ This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -- Thiago Sousa Santos Embedded Systems and Pervasive Computing Lab (Embedded) Center of Electrical Engineering and Informatics (CEEI) Federal University of Campina Grande (UFCG) Check out the all-new Messenger 9.0! Go to http://in.messenger.yahoo.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From tim.hilden at googlemail.com Tue Jan 20 17:23:37 2009 From: tim.hilden at googlemail.com (Tim Hilden) Date: Tue, 20 Jan 2009 17:23:37 +0100 Subject: [gst-devel] jpegenc only decodes one frame, then stream stops Message-ID: <1232468617.6039.53.camel@calculus> hi, I need gstreamer for the following: The frames of a video stream need to be converted to jpeg images. So I connected a filesrc-demuxer-decoder-jpegenc-identity-fakesink. The identiry's "handoff"-signal is used to read the current frame and store it to a file. The storing will be replaced with some more processing later which is not part of gstreamer. But the problem is that this stream only encodes the first frame. After that the stream is "stuck". I did not get any error messages on the bus and the q_main_loop is still running. So do I have to tell the jpegenc to encode the next frame or anything? The lack of doc and sample code is what makes programming gstreamer so hard for me. That and the fact that I don't really want to use GLib and main_loops, because it's for a QT app. But QT's phonon is way not enough for what I need. one more (simple) question: does a stream have to have a sink? Would the fakesink be neccessary in my case? Tim From tim.hilden at googlemail.com Tue Jan 20 20:36:05 2009 From: tim.hilden at googlemail.com (Tim Hilden) Date: Tue, 20 Jan 2009 20:36:05 +0100 Subject: [gst-devel] element specific bus messages In-Reply-To: <3c1737210901201016p35764100v985520ff16b47d3d@mail.gmail.com> References: <1232467772.6039.40.camel@calculus> <3c1737210901201016p35764100v985520ff16b47d3d@mail.gmail.com> Message-ID: <1232480165.6039.61.camel@calculus> well, the following code reads the bus messages: gboolean bus_callback(GstBus *bus, GstMessage *message, gpointer unused){ printf("%s: %s\n", gst_object_get_name(message->src),\ gst_message_type_get_name(message->type)); switch(GST_MESSAGE_TYPE(message)){ case GST_MESSAGE_ERROR:{ fprintf(stderr, "ERROR\n"); g_main_loop_quit(loop); break; } case GST_MESSAGE_EOS:{ printf("end-of-stream\n"); g_main_loop_quit(loop); break; } } return true; } generates the following output : demuxer: tag videosink: element videosink: element videosink: state-changed to me this shows that I rereived an element specific message from element "videosink", which in this case is a xvimagesink. now the doc for xvimagesink http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-xvimagesink.html doesn't tell me anything about a specific message (unlike the spectrum element for example). And how am I supposed to make the connection from xvimagesink to the GstXOverlay definition by just looking at the URL 3 lines up? Tim On Tue, 2009-01-20 at 10:16 -0800, Michael Smith wrote: > On Tue, Jan 20, 2009 at 8:09 AM, Tim Hilden wrote: > > hi there, > > > > playing around with the (sparsly documented) Gstreamer a couple of > > questions came to my mind. > > > > This one is about the element specific bus messages. Getting a message > > of type GST_MESSAGE_ELEMENT I wonder where I can find the "specific > > element's documentation" that is mentioned in the API doc. The online > > doc of elements like xvimagesink or jpegenc don't contain any > > information on what the element's bus messages contain. So where can I > > find that information? > > That would mean a) those elements don't have any element-specific > messages, or b) they're missing documentation for any they might send. > > In this case, they just don't have any element-specific messages > (though xvimagesink will send a "prepare-xwindow-id" message - as > documented in the GstXOverlay interface, which it implements. > > Mike > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From bilboed at gmail.com Wed Jan 28 18:17:15 2009 From: bilboed at gmail.com (Edward Hervey) Date: Wed, 28 Jan 2009 18:17:15 +0100 Subject: [gst-devel] (no subject) In-Reply-To: <595045.24291.qm@web94814.mail.in2.yahoo.com> References: <497B3E34.6050604@hora-obscura.de> <595045.24291.qm@web94814.mail.in2.yahoo.com> Message-ID: <1233163035.3853.14.camel@localhost> Don't worry fellas, I just unsubscribed him, he shouldn't cause any more harm. Move along, nothing to see here, Edward On Wed, 2009-01-28 at 22:30 +0530, sai charmboy wrote: > shut up all of u.....plz > > saicharmboy... > > > > > ______________________________________________________________________ > From: thiagoss > To: Discussion of the development of GStreamer > > Sent: Wednesday, 28 January, 2009 10:17:32 PM > Subject: Re: [gst-devel] Building failing for qtwrapper > > > > On Sat, Jan 24, 2009 at 2:13 PM, Stefan Kost > wrote: > thiagoss schrieb: > > Qtwraper building is failing (cvs head), I don't know if > this is a > > misconfiguration in my system or in Makefile, so I'm sending > this mail. > > The error output is: > > > > Making all in qtwrapper > > cd ../.. && /bin/sh > > /Users/thiagoss/Work/GStreamer/head/gst-plugins-bad/missing > --run > > automake-1.10 --gnu sys/qtwrapper/Makefile > > > configure.ac:111 : warning: > > AC_CACHE_VAL(g_support_dowhile_macros, ...): suspicious > cache-id, must > > contain _cv_ to be cached > > ../../lib/autoconf/general.m4:1974: AC_CACHE_VAL is expanded > from... > > ../../lib/autoconf/general.m4:1994: AC_CACHE_CHECK is > expanded from... > > common/m4/gst-dowhile.m4:8: AG_GST_CHECK_DOWHILE_MACROS is > expanded from... > > > configure..ac:111 : the top level > > configure.ac:143 : warning: > > AC_CACHE_VAL(GST_FIONREAD_IN_SYS_IOCTL, ...): suspicious > cache-id, must > > contain _cv_ to be cached > > m4/gst-fionread.m4:1: GST_CHECK_FIONREAD is expanded from... > > > configure.ac:143 : the top level > > configure..ac:143 : warning: > > > AC_CACHE_VAL(GST_FIONREAD_IN_SYS_FILIO, ...): suspicious > cache-id, must > > contain _cv_ to be cached > > cd ../.. && /bin/sh ./config.status sys/qtwrapper/Makefile > depfiles > > config.status: creating sys/qtwrapper/Makefile > > config.status: executing depfiles commands > > /bin/sh ../../libtool --tag=CC --tag=disable-static > --mode=compile gcc > > -DHAVE_CONFIG_H -I. -I../.. -I../../gst-libs > -I../../gst-libs > > -D_REENTRANT -I/usr/local/include/gstreamer-0.10 > > -I/opt/local/include/glib-2.0 > -I/opt/local/lib/glib-2.0/include > > -I/opt/local/include -I/opt/local/include/libxml2 -Wall > > -Wdeclaration-after-statement -Werror -fno-strict-aliasing > -g > > -D_REENTRANT -I/usr/local/include/gstreamer-0.10 > > -I/opt/local/include/glib-2.0 > -I/opt/local/lib/glib-2.0/include > > -I/opt/local/include -I/opt/local/include/libxml2 -g -O2 > -MT > > libgstqtwrapper_la-qtwrapper.lo -MD -MP -MF > > .deps/libgstqtwrapper_la-qtwrapper.Tpo -c -o > > libgstqtwrapper_la-qtwrapper.lo `test -f 'qtwrapper.c' || > echo > > './'`qtwrapper.c > > libtool: compile: gcc -DHAVE_CONFIG_H -I. -I../.. > -I../../gst-libs > > -I../../gst-libs -D_REENTRANT > -I/usr/local/include/gstreamer-0.10 > > -I/opt/local/include/glib-2.0 > -I/opt/local/lib/glib-2.0/include > > -I/opt/local/include -I/opt/local/include/libxml2 -Wall > > -Wdeclaration-after-statement -Werror -fno-strict-aliasing > -g > > -D_REENTRANT -I/usr/local/include/gstreamer-0.10 > > -I/opt/local/include/glib-2.0 > -I/opt/local/lib/glib-2.0/include > > -I/opt/local/include -I/opt/local/include/libxml2 -g -O2 -MT > > libgstqtwrapper_la-qtwrapper.lo -MD -MP -MF > > .deps/libgstqtwrapper_la-qtwrapper.Tpo -c qtwrapper.c > -fno-common -DPIC > > -o .libs/libgstqtwrapper_la-qtwrapper.o > > In file included from qtwrapper.c:49: > > qtwrapper.h:53:20: error: Movies.h: No such file or > directory > > qtwrapper.c: In function 'plugin_init': > > qtwrapper.c:58: error: 'OSErr' undeclared (first use in this > function) > > qtwrapper.c:58: error: (Each undeclared identifier is > reported only once > > qtwrapper.c:58: error: for each function it appears in.) > > qtwrapper.c:58: error: syntax error before 'status' > > qtwrapper.c:69: error: 'status' undeclared (first use in > this function) > > cc1: warnings being treated as errors > > qtwrapper.c:69: warning: implicit declaration of function > 'EnterMovies' > > make[3]: *** [libgstqtwrapper_la-qtwrapper.lo] Error 1 > > make[2]: *** [all-recursive] Error 1 > > make[1]: *** [all-recursive] Error 1 > > make: *** [all] Error 2 > > > > How to fix this? > > > It seems that configure belives you have all the needed > external libs and > headers, but in fact you don't. Do you have a Movies.h? Maybe > you can look at > config.log and check what its looking for. > > Stefan > > > # find / -name "Movies.h" > /Developer/Headers/FlatCarbon/Movies.h > /Developer/SDKs/MacOSX10.4u.sdk/Developer/Headers/CFMCarbon/Movies.h > /Developer/SDKs/MacOSX10.4u.sdk/Developer/Headers/CFMCarbon/QuickTime/Movies.h > /Developer/SDKs/MacOSX10.4u.sdk/Developer/Headers/FlatCarbon/Movies.h > /Developer/SDKs/MacOSX10.4u.sdk/System/Library/Frameworks/QuickTime.framework/Versions/A/Headers/Movies..h > /Developer/SDKs/MacOSX10.5.sdk/Developer/Headers/FlatCarbon/Movies.h > /Developer/SDKs/MacOSX10.5.sdk/System/Library/Frameworks/QuickTime.framework/Versions/A/Headers/Movies.h > /System/Library/Frameworks/QuickTime.framework/Versions/A/Headers/Movies.h > > > > Maybe it is looking in the wrong places.. > > > The config.log has this: > > > configure:30272: *** checking feature: QuickTime wrapper *** > configure:30276: *** for plug-ins: qtwrapper *** > configure:30324: checking QuickTime/Movies.h usability > configure:30341: gcc -c -g -O2 conftest.c >&5 > configure:30348: $? = 0 > configure:30362: result: yes > configure:30366: checking QuickTime/Movies.h presence > configure:30381: gcc -E conftest.c > configure:30388: $? = 0 > configure:30402: result: yes > configure:30435: checking for QuickTime/Movies.h > configure:30442: result: yes > configure:30464: *** These plugins will be built: qtwrapper > > > > > > > > > > -- > > Thiago Sousa Santos > > > > Embedded Systems and Pervasive Computing Lab (Embedded) > > Center of Electrical Engineering and Informatics (CEEI) > > Federal University of Campina Grande (UFCG) > > > > > > > > ------------------------------------------------------------------------ > > > > > ------------------------------------------------------------------------------ > > This SF.net email is sponsored by: > > SourcForge Community > > SourceForge wants to tell your story. > > http://p.sf.net/sfu/sf-spreadtheword > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > -- > Thiago Sousa Santos > > Embedded Systems and Pervasive Computing Lab (Embedded) > Center of Electrical Engineering and Informatics (CEEI) > Federal University of Campina Grande (UFCG) > > > > ______________________________________________________________________ > Add more friends to your messenger and enjoy! Invite them now. > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From msmith at xiph.org Wed Jan 28 19:08:39 2009 From: msmith at xiph.org (Michael Smith) Date: Wed, 28 Jan 2009 10:08:39 -0800 Subject: [gst-devel] Appsrc question In-Reply-To: References: Message-ID: <3c1737210901281008h6a83381by3f21ed153d94b1b6@mail.gmail.com> On Wed, Jan 28, 2009 at 8:00 AM, Darren Staples wrote: > Can you tell me please, in a video processing pipeline, is the appsrc > component used to feed video frames into the pipeline, or raw video stream > data ? Darren, If your application uses appsrc, then it's used to feed anything you want to into the pipeline - it's entirely up to the application. Mike From msmith at xiph.org Wed Jan 28 19:13:40 2009 From: msmith at xiph.org (Michael Smith) Date: Wed, 28 Jan 2009 10:13:40 -0800 Subject: [gst-devel] element specific bus messages In-Reply-To: <1232480165.6039.61.camel@calculus> References: <1232467772.6039.40.camel@calculus> <3c1737210901201016p35764100v985520ff16b47d3d@mail.gmail.com> <1232480165.6039.61.camel@calculus> Message-ID: <3c1737210901281013h4c137762qe7fd9ae4af19f722@mail.gmail.com> > > now the doc for xvimagesink > http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-xvimagesink.html > doesn't tell me anything about a specific message (unlike the spectrum > element for example). And how am I supposed to make the connection from > xvimagesink to the GstXOverlay definition by just looking at the URL 3 > lines up? The documentation pretty consistently (and sensibly) doesn't duplicate docs for the interfaces and superclasses of each element. So, from that URL, the obvious thing to do is to look at documentation for the interfaces and superclasses. In this case, that page links to GstXOverlay as one of the interfaces this element implements. Unfortunately, the link is broken. I'm not sure why; clearly that's a mistake (GstXOverlay does have documentation). I'll ask around to see if anyone knows why these links are broken. Mike From erappleman at gmail.com Wed Jan 28 22:44:04 2009 From: erappleman at gmail.com (Eric Appleman) Date: Wed, 28 Jan 2009 16:44:04 -0500 Subject: [gst-devel] libavcodec52-unstripped causes segfaults. libavcodec doesn't. In-Reply-To: <3c1737210901281008h6a83381by3f21ed153d94b1b6@mail.gmail.com> References: <3c1737210901281008h6a83381by3f21ed153d94b1b6@mail.gmail.com> Message-ID: <4980D1A4.3070504@gmail.com> Unfortunately, the unstripped is now default for Ubuntu Jaunty. Installing the vanilla version fixes the issue. If you want a gdb log, I can attach one. - Eric From julien.isorce at gmail.com Wed Jan 28 22:53:08 2009 From: julien.isorce at gmail.com (Julien Isorce) Date: Wed, 28 Jan 2009 22:53:08 +0100 Subject: [gst-devel] jpegenc only decodes one frame, then stream stops In-Reply-To: <1232468617.6039.53.camel@calculus> References: <1232468617.6039.53.camel@calculus> Message-ID: <180a127d0901281353w7cb898b6l20a25e02e462e81a@mail.gmail.com> Hi, If you do not want to use glibmainloop inside qt code see : "qt-opensource-src-4.4.3/include/QtGui/private/qguieventdispatcher_glib_p.h" The other way is to use both qtmainloop and glibmainloop, here 3 examples: http://diracvideo.org/git?p=gst-plugins-gl.git;a=tree;f=tests/examples/qt;h=925d3f5b596ceaabdbb360f69498315a09f30614;hb=42b9175bfd7179037ef4989cec0c4550811d19e0 Julien 2009/1/20 Tim Hilden > hi, > > I need gstreamer for the following: The frames of a video stream need to > be converted to jpeg images. So I connected a > filesrc-demuxer-decoder-jpegenc-identity-fakesink. > The identiry's "handoff"-signal is used to read the current frame and > store it to a file. The storing will be replaced with some more > processing later which is not part of gstreamer. But the problem is that > this stream only encodes the first frame. After that the stream is > "stuck". I did not get any error messages on the bus and the q_main_loop > is still running. So do I have to tell the jpegenc to encode the next > frame or anything? > The lack of doc and sample code is what makes programming gstreamer so > hard for me. That and the fact that I don't really want to use GLib and > main_loops, because it's for a QT app. But QT's phonon is way not enough > for what I need. > > one more (simple) question: does a stream have to have a sink? Would the > fakesink be neccessary in my case? > > Tim > > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From thaytan at noraisin.net Thu Jan 29 00:03:07 2009 From: thaytan at noraisin.net (Jan Schmidt) Date: Wed, 28 Jan 2009 23:03:07 +0000 Subject: [gst-devel] libavcodec52-unstripped causes segfaults. libavcodec doesn't. In-Reply-To: <4980D1A4.3070504@gmail.com> References: <3c1737210901281008h6a83381by3f21ed153d94b1b6@mail.gmail.com> <4980D1A4.3070504@gmail.com> Message-ID: <1233183787.31252.2.camel@fancy-ubuntu> On Wed, 2009-01-28 at 16:44 -0500, Eric Appleman wrote: > Unfortunately, the unstripped is now default for Ubuntu Jaunty. > Installing the vanilla version fixes the issue. > > If you want a gdb log, I can attach one. > Complain to Ubuntu. We don't support building gst-ffmpeg with anything but the included ffmpeg snapshot. J. -- Jan Schmidt From thaytan at noraisin.net Thu Jan 29 00:14:02 2009 From: thaytan at noraisin.net (Jan Schmidt) Date: Wed, 28 Jan 2009 23:14:02 +0000 Subject: [gst-devel] jpegenc only decodes one frame, then stream stops In-Reply-To: <1232468617.6039.53.camel@calculus> References: <1232468617.6039.53.camel@calculus> Message-ID: <1233184442.31252.9.camel@fancy-ubuntu> On Tue, 2009-01-20 at 17:23 +0100, Tim Hilden wrote: > hi, > > I need gstreamer for the following: The frames of a video stream need to > be converted to jpeg images. So I connected a > filesrc-demuxer-decoder-jpegenc-identity-fakesink. > The identiry's "handoff"-signal is used to read the current frame and > store it to a file. The storing will be replaced with some more > processing later which is not part of gstreamer. But the problem is that > this stream only encodes the first frame. After that the stream is > "stuck". I did not get any error messages on the bus and the q_main_loop > is still running. So do I have to tell the jpegenc to encode the next > frame or anything? > The lack of doc and sample code is what makes programming gstreamer so > hard for me. That and the fact that I don't really want to use GLib and > main_loops, because it's for a QT app. But QT's phonon is way not enough > for what I need. > > one more (simple) question: does a stream have to have a sink? Would the > fakesink be neccessary in my case? > jpegenc per-se encodes multiple incoming video frames into a sequence of jpeg buffers, as this pipeline shows: gst-launch filesrc location=blah.avi ! decodebin ! jpegenc ! identity ! fakesink -v I'm not sure where your problem is from there... perhaps you're not linking something properly and the pipeline is shutting down. In that case though, you should definitely be seeing an error message on the bus. Yes, your pipeline needs a sink of some sort for the buffers to end up and be disposed. The overhead isn't high. In fact though, fakesink has a handoff signal and signal-handoffs property too - so you can just connect to that directly, and forgo the identity element. Alternatively, you can also look at the appsink element, which will give you the ability to pull the buffers at your leisure, instead of having to handle them in the handoff. It might increase the minimum version of GStreamer you require though - appsink is only available in a stable form as of 0.10.22 of gst-plugins-base. See: http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-appsink.html Cheers, J. -- Jan Schmidt From dstaples at friendmts.co.uk Thu Jan 29 10:15:05 2009 From: dstaples at friendmts.co.uk (Darren Staples) Date: Thu, 29 Jan 2009 09:15:05 +0000 Subject: [gst-devel] Counting video frames Message-ID: I?ve come across a problem processing individual frames of a movie using the appsrc component. The code in my main loop looks something like this: if ( !g_queue_is_empty( GST_APP_SINK( sink )->queue )) { // Get access to raw frame buffer GstBuffer *buff = gst_app_sink_pull_buffer( GST_APP_SINK( sink )); if ( buff ) { // Get pointer to frame data guint8 *buff_data = (guint8 *)GST_BUFFER_DATA( buff ); // Do something with buffer here // Release buffer gst_buffer_unref( buff ); buff = NULL; num_frames++; } } This works, but if I run it a few times, using the same video each time, I sometimes see that the number of frames processed is lower than expected. This happens maybe once in five runs. I used a gst_app_sink_set_drop( sink , false ) to ensure that the pipeline stalls if I don?t pull buffers quickly enough. Any ideas why or where I might be dropping frames ? On a similar topic, is there a mechanism within Gstreamer for identifying the number of frames in a movie file ? Thanks ! Darren *** FRIEND MTS LIMITED EMAIL CONFIDENTIALITY *** This email message and any attachments may contain information which is confidential or privileged and is intended for the sole use of the person to whom it is addressed. If you are not the intended recipient, be aware that any disclosures, copying, distribution or use of the contents is prohibited. If you have received this email message in error, please notify our office by telephone (+44 (0)121 633 2000) or email (admin at friendmts.co.uk) immediately. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: From aniruddhachourasia at tataelxsi.co.in Thu Jan 29 13:35:47 2009 From: aniruddhachourasia at tataelxsi.co.in (Aniruddha) Date: Thu, 29 Jan 2009 18:05:47 +0530 Subject: [gst-devel] Not returning from gst_pad_push() in rtpbin element References: Message-ID: <004301c9820e$1c6369c0$3f033c0a@telxsi.com> Hi All, I am using rtpbin element of gst-plugins-farsight-0.12.9 compiled with jrtplib support on a FC6 platform. I am running the following pipeline : - "gst-launch filersrc location= ! mpeg4vidoeparse ! video/mpeg, width=352, height=288, framerate=5/1 ! rtpmp4vpay mtu=1400 ! rtpbin localport=5000 destinations=10.60.3.63:5000" No data is streamed. When i traced the rtpbin plugin, i found out that gst_pad_push() function present in the function push_packet_on_pad() in the file jrtplib_c.cpp is not returning. I also found out that gst_dynudpsink_render() function of dynudpsink element is not getting called indicating no data is being pushed. Can you tell where i am going wrong. Regards Ani ----- Original Message ----- From: To: Sent: Thursday, January 29, 2009 3:46 AM Subject: gstreamer-devel Digest, Vol 32, Issue 64 > Send gstreamer-devel mailing list submissions to > gstreamer-devel at lists.sourceforge.net > > 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 > gstreamer-devel-request at lists.sourceforge.net > > You can reach the person managing the list at > gstreamer-devel-owner at lists.sourceforge.net > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of gstreamer-devel digest..." > > > Today's Topics: > > 1. jpegenc only decodes one frame, then stream stops (Tim Hilden) > 2. Re: element specific bus messages (Tim Hilden) > 3. Re: (no subject) (Edward Hervey) > 4. Re: Appsrc question (Michael Smith) > 5. Re: element specific bus messages (Michael Smith) > 6. libavcodec52-unstripped causes segfaults. libavcodec doesn't. > (Eric Appleman) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 20 Jan 2009 17:23:37 +0100 > From: Tim Hilden > Subject: [gst-devel] jpegenc only decodes one frame, then stream stops > To: gstreamer-devel at lists.sourceforge.net > Message-ID: <1232468617.6039.53.camel at calculus> > Content-Type: text/plain > > hi, > > I need gstreamer for the following: The frames of a video stream need to > be converted to jpeg images. So I connected a > filesrc-demuxer-decoder-jpegenc-identity-fakesink. > The identiry's "handoff"-signal is used to read the current frame and > store it to a file. The storing will be replaced with some more > processing later which is not part of gstreamer. But the problem is that > this stream only encodes the first frame. After that the stream is > "stuck". I did not get any error messages on the bus and the q_main_loop > is still running. So do I have to tell the jpegenc to encode the next > frame or anything? > The lack of doc and sample code is what makes programming gstreamer so > hard for me. That and the fact that I don't really want to use GLib and > main_loops, because it's for a QT app. But QT's phonon is way not enough > for what I need. > > one more (simple) question: does a stream have to have a sink? Would the > fakesink be neccessary in my case? > > Tim > > > > > ------------------------------ > > Message: 2 > Date: Tue, 20 Jan 2009 20:36:05 +0100 > From: Tim Hilden > Subject: Re: [gst-devel] element specific bus messages > To: Discussion of the development of GStreamer > > Message-ID: <1232480165.6039.61.camel at calculus> > Content-Type: text/plain > > well, the following code reads the bus messages: > > gboolean bus_callback(GstBus *bus, GstMessage *message, gpointer > unused){ > printf("%s: %s\n", gst_object_get_name(message->src),\ > gst_message_type_get_name(message->type)); > > switch(GST_MESSAGE_TYPE(message)){ > case GST_MESSAGE_ERROR:{ > fprintf(stderr, "ERROR\n"); > g_main_loop_quit(loop); > break; > } > case GST_MESSAGE_EOS:{ > printf("end-of-stream\n"); > g_main_loop_quit(loop); > break; > } > } > > return true; > } > > generates the following output : > > demuxer: tag > videosink: element > videosink: element > videosink: state-changed > > to me this shows that I rereived an element specific message from > element "videosink", which in this case is a xvimagesink. > > now the doc for xvimagesink > http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-xvimagesink.html > doesn't tell me anything about a specific message (unlike the spectrum > element for example). And how am I supposed to make the connection from > xvimagesink to the GstXOverlay definition by just looking at the URL 3 > lines up? > > > Tim > > On Tue, 2009-01-20 at 10:16 -0800, Michael Smith wrote: >> On Tue, Jan 20, 2009 at 8:09 AM, Tim Hilden >> wrote: >> > hi there, >> > >> > playing around with the (sparsly documented) Gstreamer a couple of >> > questions came to my mind. >> > >> > This one is about the element specific bus messages. Getting a message >> > of type GST_MESSAGE_ELEMENT I wonder where I can find the "specific >> > element's documentation" that is mentioned in the API doc. The online >> > doc of elements like xvimagesink or jpegenc don't contain any >> > information on what the element's bus messages contain. So where can I >> > find that information? >> >> That would mean a) those elements don't have any element-specific >> messages, or b) they're missing documentation for any they might send. >> >> In this case, they just don't have any element-specific messages >> (though xvimagesink will send a "prepare-xwindow-id" message - as >> documented in the GstXOverlay interface, which it implements. >> >> Mike >> >> ------------------------------------------------------------------------------ >> This SF.net email is sponsored by: >> SourcForge Community >> SourceForge wants to tell your story. >> http://p.sf.net/sfu/sf-spreadtheword >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------ > > Message: 3 > Date: Wed, 28 Jan 2009 18:17:15 +0100 > From: Edward Hervey > Subject: Re: [gst-devel] (no subject) > To: Discussion of the development of GStreamer > > Message-ID: <1233163035.3853.14.camel at localhost> > Content-Type: text/plain > > Don't worry fellas, I just unsubscribed him, he shouldn't cause any more > harm. > > Move along, nothing to see here, > > Edward > > > On Wed, 2009-01-28 at 22:30 +0530, sai charmboy wrote: >> shut up all of u.....plz >> >> saicharmboy... >> >> >> >> >> ______________________________________________________________________ >> From: thiagoss >> To: Discussion of the development of GStreamer >> >> Sent: Wednesday, 28 January, 2009 10:17:32 PM >> Subject: Re: [gst-devel] Building failing for qtwrapper >> >> >> >> On Sat, Jan 24, 2009 at 2:13 PM, Stefan Kost >> wrote: >> thiagoss schrieb: >> > Qtwraper building is failing (cvs head), I don't know if >> this is a >> > misconfiguration in my system or in Makefile, so I'm sending >> this mail. >> > The error output is: >> > >> > Making all in qtwrapper >> > cd ../.. && /bin/sh >> > /Users/thiagoss/Work/GStreamer/head/gst-plugins-bad/missing >> --run >> > automake-1.10 --gnu sys/qtwrapper/Makefile >> >> > configure.ac:111 : warning: >> > AC_CACHE_VAL(g_support_dowhile_macros, ...): suspicious >> cache-id, must >> > contain _cv_ to be cached >> > ../../lib/autoconf/general.m4:1974: AC_CACHE_VAL is expanded >> from... >> > ../../lib/autoconf/general.m4:1994: AC_CACHE_CHECK is >> expanded from... >> > common/m4/gst-dowhile.m4:8: AG_GST_CHECK_DOWHILE_MACROS is >> expanded from... >> >> > configure..ac:111 : the top level >> > configure.ac:143 : warning: >> > AC_CACHE_VAL(GST_FIONREAD_IN_SYS_IOCTL, ...): suspicious >> cache-id, must >> > contain _cv_ to be cached >> > m4/gst-fionread.m4:1: GST_CHECK_FIONREAD is expanded from... >> >> > configure.ac:143 : the top level >> > configure..ac:143 : warning: >> >> > AC_CACHE_VAL(GST_FIONREAD_IN_SYS_FILIO, ...): suspicious >> cache-id, must >> > contain _cv_ to be cached >> > cd ../.. && /bin/sh ./config.status sys/qtwrapper/Makefile >> depfiles >> > config.status: creating sys/qtwrapper/Makefile >> > config.status: executing depfiles commands >> > /bin/sh ../../libtool --tag=CC --tag=disable-static >> --mode=compile gcc >> > -DHAVE_CONFIG_H -I. -I../.. -I../../gst-libs >> -I../../gst-libs >> > -D_REENTRANT -I/usr/local/include/gstreamer-0.10 >> > -I/opt/local/include/glib-2.0 >> -I/opt/local/lib/glib-2.0/include >> > -I/opt/local/include -I/opt/local/include/libxml2 -Wall >> > -Wdeclaration-after-statement -Werror -fno-strict-aliasing >> -g >> > -D_REENTRANT -I/usr/local/include/gstreamer-0.10 >> > -I/opt/local/include/glib-2.0 >> -I/opt/local/lib/glib-2.0/include >> > -I/opt/local/include -I/opt/local/include/libxml2 -g -O2 >> -MT >> > libgstqtwrapper_la-qtwrapper.lo -MD -MP -MF >> > .deps/libgstqtwrapper_la-qtwrapper.Tpo -c -o >> > libgstqtwrapper_la-qtwrapper.lo `test -f 'qtwrapper.c' || >> echo >> > './'`qtwrapper.c >> > libtool: compile: gcc -DHAVE_CONFIG_H -I. -I../.. >> -I../../gst-libs >> > -I../../gst-libs -D_REENTRANT >> -I/usr/local/include/gstreamer-0.10 >> > -I/opt/local/include/glib-2.0 >> -I/opt/local/lib/glib-2.0/include >> > -I/opt/local/include -I/opt/local/include/libxml2 -Wall >> > -Wdeclaration-after-statement -Werror -fno-strict-aliasing >> -g >> > -D_REENTRANT -I/usr/local/include/gstreamer-0.10 >> > -I/opt/local/include/glib-2.0 >> -I/opt/local/lib/glib-2.0/include >> > -I/opt/local/include -I/opt/local/include/libxml2 -g -O2 -MT >> > libgstqtwrapper_la-qtwrapper.lo -MD -MP -MF >> > .deps/libgstqtwrapper_la-qtwrapper.Tpo -c qtwrapper.c >> -fno-common -DPIC >> > -o .libs/libgstqtwrapper_la-qtwrapper.o >> > In file included from qtwrapper.c:49: >> > qtwrapper.h:53:20: error: Movies.h: No such file or >> directory >> > qtwrapper.c: In function 'plugin_init': >> > qtwrapper.c:58: error: 'OSErr' undeclared (first use in this >> function) >> > qtwrapper.c:58: error: (Each undeclared identifier is >> reported only once >> > qtwrapper.c:58: error: for each function it appears in.) >> > qtwrapper.c:58: error: syntax error before 'status' >> > qtwrapper.c:69: error: 'status' undeclared (first use in >> this function) >> > cc1: warnings being treated as errors >> > qtwrapper.c:69: warning: implicit declaration of function >> 'EnterMovies' >> > make[3]: *** [libgstqtwrapper_la-qtwrapper.lo] Error 1 >> > make[2]: *** [all-recursive] Error 1 >> > make[1]: *** [all-recursive] Error 1 >> > make: *** [all] Error 2 >> > >> > How to fix this? >> >> >> It seems that configure belives you have all the needed >> external libs and >> headers, but in fact you don't. Do you have a Movies.h? Maybe >> you can look at >> config.log and check what its looking for. >> >> Stefan >> >> >> # find / -name "Movies.h" >> /Developer/Headers/FlatCarbon/Movies.h >> /Developer/SDKs/MacOSX10.4u.sdk/Developer/Headers/CFMCarbon/Movies.h >> /Developer/SDKs/MacOSX10.4u.sdk/Developer/Headers/CFMCarbon/QuickTime/Movies.h >> /Developer/SDKs/MacOSX10.4u.sdk/Developer/Headers/FlatCarbon/Movies.h >> /Developer/SDKs/MacOSX10.4u.sdk/System/Library/Frameworks/QuickTime.framework/Versions/A/Headers/Movies..h >> /Developer/SDKs/MacOSX10.5.sdk/Developer/Headers/FlatCarbon/Movies.h >> /Developer/SDKs/MacOSX10.5.sdk/System/Library/Frameworks/QuickTime.framework/Versions/A/Headers/Movies.h >> /System/Library/Frameworks/QuickTime.framework/Versions/A/Headers/Movies.h >> >> >> >> Maybe it is looking in the wrong places.. >> >> >> The config.log has this: >> >> >> configure:30272: *** checking feature: QuickTime wrapper *** >> configure:30276: *** for plug-ins: qtwrapper *** >> configure:30324: checking QuickTime/Movies.h usability >> configure:30341: gcc -c -g -O2 conftest.c >&5 >> configure:30348: $? = 0 >> configure:30362: result: yes >> configure:30366: checking QuickTime/Movies.h presence >> configure:30381: gcc -E conftest.c >> configure:30388: $? = 0 >> configure:30402: result: yes >> configure:30435: checking for QuickTime/Movies.h >> configure:30442: result: yes >> configure:30464: *** These plugins will be built: qtwrapper >> >> >> >> >> >> >> > >> > -- >> > Thiago Sousa Santos >> > >> > Embedded Systems and Pervasive Computing Lab (Embedded) >> > Center of Electrical Engineering and Informatics (CEEI) >> > Federal University of Campina Grande (UFCG) >> > >> > >> >> > >> ------------------------------------------------------------------------ >> > >> > >> ------------------------------------------------------------------------------ >> > This SF.net email is sponsored by: >> > SourcForge Community >> > SourceForge wants to tell your story. >> > http://p.sf.net/sfu/sf-spreadtheword >> > >> > >> > >> ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > gstreamer-devel mailing list >> > gstreamer-devel at lists.sourceforge.net >> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> ------------------------------------------------------------------------------ >> This SF.net email is sponsored by: >> SourcForge Community >> SourceForge wants to tell your story. >> http://p.sf.net/sfu/sf-spreadtheword >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> -- >> Thiago Sousa Santos >> >> Embedded Systems and Pervasive Computing Lab (Embedded) >> Center of Electrical Engineering and Informatics (CEEI) >> Federal University of Campina Grande (UFCG) >> >> >> >> ______________________________________________________________________ >> Add more friends to your messenger and enjoy! Invite them now. >> ------------------------------------------------------------------------------ >> This SF.net email is sponsored by: >> SourcForge Community >> SourceForge wants to tell your story. >> http://p.sf.net/sfu/sf-spreadtheword >> _______________________________________________ gstreamer-devel mailing >> list gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------ > > Message: 4 > Date: Wed, 28 Jan 2009 10:08:39 -0800 > From: Michael Smith > Subject: Re: [gst-devel] Appsrc question > To: Discussion of the development of GStreamer > > Message-ID: > <3c1737210901281008h6a83381by3f21ed153d94b1b6 at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > On Wed, Jan 28, 2009 at 8:00 AM, Darren Staples > wrote: >> Can you tell me please, in a video processing pipeline, is the appsrc >> component used to feed video frames into the pipeline, or raw video >> stream >> data ? > > Darren, > > If your application uses appsrc, then it's used to feed anything you > want to into the pipeline - it's entirely up to the application. > > Mike > > > > ------------------------------ > > Message: 5 > Date: Wed, 28 Jan 2009 10:13:40 -0800 > From: Michael Smith > Subject: Re: [gst-devel] element specific bus messages > To: Discussion of the development of GStreamer > > Message-ID: > <3c1737210901281013h4c137762qe7fd9ae4af19f722 at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > >> >> now the doc for xvimagesink >> http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-xvimagesink.html >> doesn't tell me anything about a specific message (unlike the spectrum >> element for example). And how am I supposed to make the connection from >> xvimagesink to the GstXOverlay definition by just looking at the URL 3 >> lines up? > > The documentation pretty consistently (and sensibly) doesn't duplicate > docs for the interfaces and superclasses of each element. > > So, from that URL, the obvious thing to do is to look at documentation > for the interfaces and superclasses. In this case, that page links to > GstXOverlay as one of the interfaces this element implements. > > Unfortunately, the link is broken. I'm not sure why; clearly that's a > mistake (GstXOverlay does have documentation). I'll ask around to see > if anyone knows why these links are broken. > > Mike > > > > ------------------------------ > > Message: 6 > Date: Wed, 28 Jan 2009 16:44:04 -0500 > From: Eric Appleman > Subject: [gst-devel] libavcodec52-unstripped causes segfaults. > libavcodec doesn't. > To: Discussion of the development of GStreamer > > Message-ID: <4980D1A4.3070504 at gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Unfortunately, the unstripped is now default for Ubuntu Jaunty. > Installing the vanilla version fixes the issue. > > If you want a gdb log, I can attach one. > > - Eric > > > > > ------------------------------ > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > > ------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > End of gstreamer-devel Digest, Vol 32, Issue 64 > *********************************************** The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. From ylatuya at gmail.com Thu Jan 29 16:36:34 2009 From: ylatuya at gmail.com (Andoni Morales) Date: Thu, 29 Jan 2009 16:36:34 +0100 Subject: [gst-devel] Building functional GStreamer binaries for Windows Message-ID: <772db3280901290736n713688e0hb772245be182831f@mail.gmail.com> Hi, I recently imported to my svn repository all the changes we made to OAbuild in order to compile functional GStreamer binaries on Windows based on the OAbuild method. I know that OAbuild is changing to OAHbuild, but until all the migration is done I think this method can be very usefull for those who want to compile their own binaries on Windows. The changes we made so far are: ?Added ffmpeg plugin with full codec support .Glib 2.18 .Added volume plugin .Added libmad plugin .Added asfdemux plugin. .Added lame plugin ?Removed Direct Show wrapper and use the ffmepg plugin for all the decoding stuff. ?Added xvid plugin. ?Added the mpeg2decoder. ?Added mpegstream plugin. .Added resindvd plugin. .Added libfaad plugin. .Added libfaac plugin. .Added libdvdspu plugin. .Adde liba52 plugin. .Added mpeg4parse plugin. .Added qtdemux plugin. You can get a working copy of the reposiroty with the following command: svn checkout *https://forja.rediris.es/svn/csl-longomatch/trunk/** gstreamer_oabuild To compile the Direct Show plugins follow the instructions at http://people.collabora.co.uk/~oleavr/OABuild/ and change in build\vsprops\Common.vsprops the following macro with your installation path: The first build usually throws 2 errors for 2 plugins which are solved builden agian the whole solution (not Rebuild, just Build) I encourage every one to add new plugins as many of them are still missing. You can found precompiled binaries with this method at: http://forja.rediris.es/frs/?group_id=96 Regards, Andoni Morales * -------------- next part -------------- An HTML attachment was scrubbed... URL: From ensonic at hora-obscura.de Thu Jan 29 16:57:49 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Thu, 29 Jan 2009 17:57:49 +0200 Subject: [gst-devel] element specific bus messages In-Reply-To: <3c1737210901281013h4c137762qe7fd9ae4af19f722@mail.gmail.com> References: <1232467772.6039.40.camel@calculus> <3c1737210901201016p35764100v985520ff16b47d3d@mail.gmail.com> <1232480165.6039.61.camel@calculus> <3c1737210901281013h4c137762qe7fd9ae4af19f722@mail.gmail.com> Message-ID: <4981D1FD.2000007@hora-obscura.de> Michael Smith schrieb: >> now the doc for xvimagesink >> http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-xvimagesink.html >> doesn't tell me anything about a specific message (unlike the spectrum >> element for example). And how am I supposed to make the connection from >> xvimagesink to the GstXOverlay definition by just looking at the URL 3 >> lines up? >> > > The documentation pretty consistently (and sensibly) doesn't duplicate > docs for the interfaces and superclasses of each element. > > So, from that URL, the obvious thing to do is to look at documentation > for the interfaces and superclasses. In this case, that page links to > GstXOverlay as one of the interfaces this element implements. > > Unfortunately, the link is broken. I'm not sure why; clearly that's a > mistake (GstXOverlay does have documentation). I'll ask around to see > if anyone knows why these links are broken. > > Mike > > They are uploaded wrongly. It needs a gtkdoc-rebase --online and then uploading. Stefan From brijesh.ksingh at gmail.com Fri Jan 30 00:30:50 2009 From: brijesh.ksingh at gmail.com (Brijesh Singh) Date: Thu, 29 Jan 2009 17:30:50 -0600 Subject: [gst-devel] Possible bug in v4l2src element - gst-plugins-good-0.10.10 Message-ID: <1b712b840901291530l4021665ajfba708c7a140da51@mail.gmail.com> Hello, I was porting v4l2src element on TI Davinci platform and found a possible bug in v4l2src element. v4l2src creates a list of supported norms by using VIDIOC_ENUMSTD ioctl and then it calls gst_v4l2_get_norm() to get the current norms and finally compares "if (norm == GST_V4L2_TUNER_NORM (item->data)->index)" to return the select norm object/ This comparison will not always return correct standard probed via the driver. because its possible that driver returns the subset of a known standard instead of the real-standard. e.g NTSC instead of NTSC_M. see v4l2 spec on why: http://www.linuxtv.org/downloads/video4linux/API/V4L2_API/spec-single/v4l2.html#STANDARD The spec has a example showing how the comparison is done. A very small patch to fix this: diff -uNr orig/gst-plugins-good-0.10.10/sys/v4l2/gstv4l2tuner.c gst-plugins-good-0.10.10/sys/v4l2/gstv4l2tuner.c --- orig/gst-plugins-good-0.10.10/sys/v4l2/gstv4l2tuner.c 2007-07-18 06:42:33.000000000 -0500 +++ gst-plugins-good-0.10.10/sys/v4l2/gstv4l2tuner.c 2009-01-29 16:11:21.000000000 -0600 @@ -244,7 +244,7 @@ gst_v4l2_get_norm (v4l2object, &norm); for (item = v4l2object->norms; item != NULL; item = item->next) { - if (norm == GST_V4L2_TUNER_NORM (item->data)->index) + if (norm & GST_V4L2_TUNER_NORM (item->data)->index) return (GstTunerNorm *) item->data; } Please correct me if i'm wrong. Thanks, Brijesh Singh From pcarlson at kgisystems.com Thu Jan 29 22:59:13 2009 From: pcarlson at kgisystems.com (Peter Carlson) Date: Thu, 29 Jan 2009 13:59:13 -0800 Subject: [gst-devel] error compiling Message-ID: <003001c9825c$d2c137d0$7843a770$@com> I am trying to compile gstreamer on redhat 9 (with mostly updated libraries) I have tried both 0.10.11 and the most current version. In 0.10.11 I get an error /usr/local/share/bison/yacc.c:37: /usr/bin/m4: ERROR: Recursion limit of 250 exceeded, use -L to change it /usr/local/bin/binson: subsidiary program `/usr/bin/mb' failed (exit status 1) Not sure what to do next. Peter From jyoti.d at allaboutif.com Fri Jan 30 10:01:50 2009 From: jyoti.d at allaboutif.com (Jyoti D) Date: Fri, 30 Jan 2009 14:31:50 +0530 Subject: [gst-devel] x264enc Message-ID: Hi all, I am encoding a video file into h264 using x264enc and playing the file. But the file doesn't play smoothly it stops very often while playing. I tried setting some properties on x264enc but I didn't see any difference. I am using the following pipeline could someone help me. gst-launch uridecodebin uri=file:/// ! x264enc ! ffdec_h264 ! ffmpegcolorspace ! xvimagesink Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From daniel.j.laird at nxp.com Fri Jan 30 10:29:13 2009 From: daniel.j.laird at nxp.com (Daniel James Laird) Date: Fri, 30 Jan 2009 10:29:13 +0100 Subject: [gst-devel] Writing a Video Sink and using hardware decoders. Message-ID: I am new to gstreamer and am writing my first plugin. The hardware I am running on is the NXP PNX8X3X hardware. This has an embedded video decoder (most usual video codecs) and a closed video pipe. This means I am looking to write a gstreamer plugin that allows the following: FileSrc- > Demux (qtdemux or other) -> My plugin. This would seem to suggest that my plugin is a Video Sink, however my plugin is also a Codec and Decoder as well. Is this possible? Or do I need to split the plugin into a Codec/Decoder and a Video Sink? Many thanks Daniel Laird -------------- next part -------------- An HTML attachment was scrubbed... URL: From hayden at heydan.co.uk Fri Jan 30 11:02:07 2009 From: hayden at heydan.co.uk (Hayden Andrews) Date: Fri, 30 Jan 2009 17:02:07 +0700 Subject: [gst-devel] Saving a live stream In-Reply-To: <200901281625.43599.arnout@mind.be> References: <497AD842.8040706@heydan.co.uk> <200901281625.43599.arnout@mind.be> Message-ID: <4982D01F.9080601@heydan.co.uk> Arnout Vandecappelle wrote: > On Saturday 24 January 2009 09:58:42 Hayden Andrews wrote: > >> First of all, I wrote a quick Perl script which reads directly from the >> appropriate HTTP port (an Icecast Server) and writes to a file. >> Everything looks fine with this file except .. >> 1) Suppose we record from 40 mins to 1:20 of the live stream. When we >> play the file, the current time starts at the time we started recording >> from rather than from 00:00. Although the run time is correct (40 mins), >> the current time progresses from 40mins to 1:20. >> > > That's because the buffer timestamps are retained. You have to re-stamp the > buffers. I ended up writing an element to do exactly that: > http://bugzilla.gnome.org/show_bug.cgi?id=561224 - that bugzilla entry also > contains a remark on how it should be done by setting base_time, but I don't > think that will work. As discussed earlier on the list (see > http://thread.gmane.org/gmane.comp.video.gstreamer.devel/22806 ), you can > also try an identity with single-segment=true, although I also doubt that > will work :-). > > In any case, you'll need to do demuxing and remuxing to get the timestamps in > your output file. So your pipeline will look like this: > > filesrc location=... ! oggdemux ! identity single-segment=true \ > ! oggmux ! filesink location=... > > > Of course, you'll encounter additional problems with audio: you'll need to > split the pipeline and add queue's, and I'm pretty sure something will go > wrong with the synchronisation between the timestamps of the audio and video > channel. > I'm going to try the 'identity' idea as your element is not yet in the build of GStreamer that is on the production machine. I'm not filled with confidence when you suggest that something else is bound to go wrong! Is this not a fairly standard scenario? We are just trying to save portion of a live stream to a file that can then be edited easily, and played with reasonable timings. Anyway, I will try the new script with an identity element for each of the audio/video streams. >> I figure that since a live stream has an unknown length, and since it is >> not possible to seek when viewing a live stream, that there is missing >> info in the live stream that needs to be inserted while saving. So, I >> also figure that it would be easy to insert the relevant requirements >> while saving a live stream to file..... >> > Indeed, the output file doesn't contain an index, which means the editor > needs to scan the entire file in order to seek correctly. The index is only > written when the muxer receives an EOS event. Unfortunately, as far as I > know, there is no way to force an EOS on a stream started with gst-launch. > How do you select the part you want using GStreamer anyway? > I'm using the CPAN GStreamer module, not gst-launch. We are currently kicking off the script via a web page. So, the person, who is videoing/streaming the event, at the require time, hits 'record stream' on a web page which kicks off the Perl script. When they want to stop recording, they press 'Stop Recording' which sends a signal to the Perl Script. When the script receives the signal, it calls quit() on the Glib::MainLoop object. I guess that I should be sending this EOS to the pipe before calling quit() ??? How do I go about doing that? Thanks heaps for your help, so far, Arnout. Hayden -------------- next part -------------- An HTML attachment was scrubbed... URL: From thomas at apestaart.org Fri Jan 30 10:42:22 2009 From: thomas at apestaart.org (Thomas Vander Stichele) Date: Fri, 30 Jan 2009 10:42:22 +0100 Subject: [gst-devel] new-segment events on decoders Message-ID: <1233308542.30298.0.camel@level.fluendo.lan> Hi everyone, I'm writing a simple application to checksum lossless CD rips and compare them against a database of known-good rips. I need to be able to get all decoded samples between two offsets. I'm assuming that the right way to do this using GStreamer is to construct a decoding pipeline with appsink, and send a new-segment event to appsink requesting the bytes I'm interested in. However, this doesn't work atm. For starters, it crashes with flacdec. That was easy enough to work around, but after that it still doesn't really work. So, my questions: - is the approach the right one ? - why do decoders not handle NEWSEGMENT events in their src_event handler ? (I checked flacdec, vorbisdec, and mad) - why does flacdec use gst_pad_event_default as the default handler, while others seem to just push the event to their sinkpad ? Thomas From bilboed at gmail.com Fri Jan 30 11:18:34 2009 From: bilboed at gmail.com (Edward Hervey) Date: Fri, 30 Jan 2009 11:18:34 +0100 Subject: [gst-devel] new-segment events on decoders In-Reply-To: <1233308542.30298.0.camel@level.fluendo.lan> References: <1233308542.30298.0.camel@level.fluendo.lan> Message-ID: <1233310714.3853.22.camel@localhost> NEW_SEGMENT is a downstream event. Just do a seek for the requested interval on the pipeline. On Fri, 2009-01-30 at 10:42 +0100, Thomas Vander Stichele wrote: > Hi everyone, > > I'm writing a simple application to checksum lossless CD rips and > compare them against a database of known-good rips. > > I need to be able to get all decoded samples between two offsets. > > I'm assuming that the right way to do this using GStreamer is to > construct a decoding pipeline with appsink, and send a new-segment event > to appsink requesting the bytes I'm interested in. > > However, this doesn't work atm. For starters, it crashes with flacdec. > That was easy enough to work around, but after that it still doesn't > really work. > > So, my questions: > > - is the approach the right one ? > - why do decoders not handle NEWSEGMENT events in their src_event > handler ? (I checked flacdec, vorbisdec, and mad) > - why does flacdec use gst_pad_event_default as the default handler, > while others seem to just push the event to their sinkpad ? > > Thomas > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From bilboed at gmail.com Fri Jan 30 11:25:04 2009 From: bilboed at gmail.com (Edward Hervey) Date: Fri, 30 Jan 2009 11:25:04 +0100 Subject: [gst-devel] Writing a Video Sink and using hardware decoders. In-Reply-To: References: Message-ID: <1233311104.3853.28.camel@localhost> Hi, On Fri, 2009-01-30 at 10:29 +0100, Daniel James Laird wrote: > I am new to gstreamer and am writing my first plugin. > > The hardware I am running on is the NXP PNX8X3X hardware. This has an > embedded video decoder (most usual video codecs) and a closed video > pipe. > > > > This means I am looking to write a gstreamer plugin that allows the > following: > > FileSrc- > Demux (qtdemux or other) -> My plugin. > > > > This would seem to suggest that my plugin is a Video Sink, however my > plugin is also a Codec and Decoder as well. Is this possible? > > Or do I need to split the plugin into a Codec/Decoder and a Video > Sink? It would be better to have separate decoders and videosinks, since this would allow a greater range of gstreamer-based application to run on your device, or to insert elements between your decoder and sink. You can still write a non-raw sink using GstBaseSink. The only thing you will have to properly handle is the synchronization/clocking. I don't think we have non-raw sinks in gst-plugins- though to serve as examples. > > > > Many thanks > > Daniel Laird > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From t.i.m at zen.co.uk Fri Jan 30 11:27:12 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Fri, 30 Jan 2009 10:27:12 +0000 Subject: [gst-devel] error compiling In-Reply-To: <003001c9825c$d2c137d0$7843a770$@com> References: <003001c9825c$d2c137d0$7843a770$@com> Message-ID: <1233311232.23330.1.camel@mini.centricular.net> On Thu, 2009-01-29 at 13:59 -0800, Peter Carlson wrote: > I am trying to compile gstreamer on redhat 9 (with mostly updated libraries) > I have tried both 0.10.11 and the most current version. > In 0.10.11 I get an error > /usr/local/share/bison/yacc.c:37: /usr/bin/m4: ERROR: Recursion limit of 250 > exceeded, use -L to change it > /usr/local/bin/binson: subsidiary program `/usr/bin/mb' failed (exit status > 1) What are you trying to compile exactly? The most current version of GStreamer (core/base) is 0.10.22. Cheers -Tim From bisht.sudarshan at gmail.com Fri Jan 30 11:37:03 2009 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Fri, 30 Jan 2009 16:07:03 +0530 Subject: [gst-devel] x264enc In-Reply-To: References: Message-ID: <785339900901300237x39767922v3fb3d8ee2e7c84a1@mail.gmail.com> Hi , Use queue in between encoder and decoder. On Fri, Jan 30, 2009 at 2:31 PM, Jyoti D wrote: > Hi all, > > I am encoding a video file into h264 using x264enc and playing the file. > But the file doesn't play smoothly it stops very often while playing. I > tried setting some properties on x264enc but I didn't see any difference. > I am using the following pipeline could someone help me. > > gst-launch uridecodebin uri=file:/// ! x264enc ! ffdec_h264 ! > ffmpegcolorspace ! xvimagesink > > > Thanks > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From daniel.j.laird at nxp.com Fri Jan 30 11:45:40 2009 From: daniel.j.laird at nxp.com (Daniel James Laird) Date: Fri, 30 Jan 2009 11:45:40 +0100 Subject: [gst-devel] [gst-embedded] Writing a Video Sink and using hardware decoders. In-Reply-To: <1233311104.3853.28.camel@localhost> References: <1233311104.3853.28.camel@localhost> Message-ID: Many thanks for the reply, we are having similar discussions internally as well. My question back would be Could I do the following? filesrc -> demux -> My plugin (Codec/Decoder) -> FakeSink This would mean I am more like the standard model. However the Fakesink does very little just satisfies the standard gstreamer model. But it means that the closed video pipe (that's handles avsync etc, hides all datapointers etc is all left as it is). I just feed my Codec/Decoder plugin with encoded video (write to the device (+ metadata like timestamps)) and everytime I decode a frame I output something to keep the FakeSink (effectively /dev/null) happy. Daniel Laird -----Original Message----- From: Edward Hervey [mailto:bilboed at gmail.com] Sent: 2009 Jan 30 10:25 To: Discussion of the development of GStreamer Cc: gstreamer-embedded at lists.sourceforge.net Subject: Re: [gst-embedded] [gst-devel] Writing a Video Sink and using hardware decoders. Hi, On Fri, 2009-01-30 at 10:29 +0100, Daniel James Laird wrote: > I am new to gstreamer and am writing my first plugin. > > The hardware I am running on is the NXP PNX8X3X hardware. This has an > embedded video decoder (most usual video codecs) and a closed video > pipe. > > > > This means I am looking to write a gstreamer plugin that allows the > following: > > FileSrc- > Demux (qtdemux or other) -> My plugin. > > > > This would seem to suggest that my plugin is a Video Sink, however my > plugin is also a Codec and Decoder as well. Is this possible? > > Or do I need to split the plugin into a Codec/Decoder and a Video > Sink? It would be better to have separate decoders and videosinks, since this would allow a greater range of gstreamer-based application to run on your device, or to insert elements between your decoder and sink. You can still write a non-raw sink using GstBaseSink. The only thing you will have to properly handle is the synchronization/clocking. I don't think we have non-raw sinks in gst-plugins- though to serve as examples. > > > > Many thanks > > Daniel Laird > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel ------------------------------------------------------------------------------ This SF.net email is sponsored by: SourcForge Community SourceForge wants to tell your story. http://p.sf.net/sfu/sf-spreadtheword _______________________________________________ Gstreamer-embedded mailing list Gstreamer-embedded at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-embedded From t.i.m at zen.co.uk Fri Jan 30 12:03:56 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Fri, 30 Jan 2009 11:03:56 +0000 Subject: [gst-devel] Possible bug in v4l2src element - gst-plugins-good-0.10.10 In-Reply-To: <1b712b840901291530l4021665ajfba708c7a140da51@mail.gmail.com> References: <1b712b840901291530l4021665ajfba708c7a140da51@mail.gmail.com> Message-ID: <1233313436.23330.5.camel@mini.centricular.net> On Thu, 2009-01-29 at 17:30 -0600, Brijesh Singh wrote: > Hello, > > I was porting v4l2src element on TI Davinci platform and found a > possible bug in v4l2src element. Could you file a bug in bugzilla please? http://gstreamer.freedesktop.org/bugs/ Cheers -Tim From maximlevitsky at gmail.com Fri Jan 30 12:56:41 2009 From: maximlevitsky at gmail.com (Maxim Levitsky) Date: Fri, 30 Jan 2009 13:56:41 +0200 Subject: [gst-devel] Few questions about status of gstreamer In-Reply-To: <772db3280901261533i2433fe69y5737fc4519ee7c65@mail.gmail.com> References: <1232490570.15685.35.camel@maxim-laptop> <772db3280901261533i2433fe69y5737fc4519ee7c65@mail.gmail.com> Message-ID: <1233316601.14752.32.camel@maxim-laptop> On Tue, 2009-01-27 at 00:33 +0100, Andoni Morales wrote: > PS: do you know whenever gstreamer can add an index to avi > file? > > I use this command to record a live asf stream: > > > > #source description > > export SOURCE_PROTOCOL=mmssrc > > export SOURCE_DEMUXER=asfdemux > > export SOURCE_AUDIO_CODEC=ffdec_wmav2 > > export SOURCE_VIDEO_CODEC=ffdec_wmv3 > > > > #destanation description > > export DEST_PROTOCOL=filesink > > export DEST_MUXER=avimux > > export DEST_AUDIO_CODEC=lame > > export DEST_VIDEO_CODEC=xvidenc > > > gst-launch > \ > > $SOURCE_PROTOCOL location=$SOURCE_LOCATION ! queue ! > $SOURCE_DEMUXER name=demuxer \ > > > \ > > demuxer. ! queue ! $SOURCE_VIDEO_CODEC ! videorate ! > postproc_default ! tee name=raw_video \ > > demuxer. ! queue ! $SOURCE_AUDIO_CODEC ! tee > name=raw_audio \ > > > \ > > raw_video. ! queue leaky=1 ! $DEST_VIDEO_CODEC ! > queue ! $DEST_MUXER name=muxer \ > > raw_audio. ! queue leaky=1 ! $DEST_AUDIO_CODEC ! > queue ! muxer. \ > > > \ > > raw_video. ! queue ! gconfvideosink > \ > > raw_audio. ! queue ! gconfaudiosink > \ > > > \ > > muxer. ! $DEST_PROTOCOL location=$DEST_LOCATION > \ > > It works, but resulting file can be seeking to non key-frames, > which > obliviously make seeking very unpleasant. > (since muxers are limited to such small number of codecs, > can't they > know which frames are key and which not ?) > > You just have to fire an EOS event to finalize the file and build the > index > Is there a way to do so with gst-launch? (I guess no) I write an app using this pipeline someday. Best regards, Maxim Levitsky From maximlevitsky at gmail.com Fri Jan 30 13:32:08 2009 From: maximlevitsky at gmail.com (Maxim Levitsky) Date: Fri, 30 Jan 2009 14:32:08 +0200 Subject: [gst-devel] Few questions about status of gstreamer In-Reply-To: <497E3579.1070808@hora-obscura.de> References: <1232490570.15685.35.camel@maxim-laptop> <497E3579.1070808@hora-obscura.de> Message-ID: <1233318728.14752.68.camel@maxim-laptop> On Tue, 2009-01-27 at 00:13 +0200, Stefan Kost wrote: > Maxim Levitsky schrieb: > > Hi, > > > > I was tinkering with gstreamer lately, and I find it qute nice platform. > > > > I would like to know where current development is headed, and what to > > expect. > > > > While the concept of gstreamer is great, the current implementation > > isn't that great, I would like to point at following things: > > > > - web live streams playback is very bad compared even to other linux > > media players, as a test case visit any web site that offers list of > > live web channels, and you will be able to watch maybe 8% og them or so, > > this is very sad. I know that there are property features, but many > > streams are yet playable in vlc or mplayer. > > I don't think its that bad, but please file bugs. Thats the only way to improve > it. Its helpful if you can try to narrow down whats causing the failure, like if > you can cluster the features that cause problems. I will do so when I have more free time, but a website with plain links, that would reflect current status of gstreamer would be very nice, also somebody with windows can test same links there, to weed out dead links (or a windows in a VM that is) > > > > I wish there were a general database of links to such streams that will > > be visited by gstreamer/totem developers as this will encourage them to > > fix support for that. > > Also such database will ease the regression testing. > > ( for example ms switched to asf over rtsp, and I see some streams that > > use it, but gstreamer will just fail as it will think it is a mms stream > > also support for wmv over rtsp isn't in gstreamer, patches are aviable, > > but it doesn't seem that they will be mereged soon. sad. I have no free > > time yet ( might have some in near future ) to do that myself. > > > > ( note that I am speaking about web sites that offer list of live > > streams, as they provide direct links without fancy media player > > testing, this rules out web page incompatibilities due to sites using > > windows only player controls) > > > > > > - gstreamer elements are far from beeng generic: > > look at that for example : > > > > maxim at maxim-laptop:~$ gst-inspect xvidenc | grep video > > Long name: XviD video encoder > > video/x-raw-yuv > > video/x-raw-rgb > > video/x-raw-rgb > > video/x-raw-rgb > > video/x-raw-rgb > > video/x-raw-rgb > > video/x-raw-rgb > > video/x-raw-rgb > > video/x-xvid > > > > as you see it supports only 'video/x-xvid' on output, but why, isn't > > xvid output plain mpeg-4 ? > > > > on the other hand > > > > maxim at maxim-laptop:~$ gst-inspect ffmux_mp4 | grep video > > video/quicktime > > SINK template: 'video_%d' > > video/mpeg > > video/x-divx > > video/x-h264 > > > > this is just one of may examples. > > Again please file bugs. You are right we have issues like this. The sheer amount > of plugins requires quite some effort on maintenance. Of course bug with patches > are even nicer. I will soon, but can't mixers just take anything inside? Why they have to be picky? > > > > > in other words currently muxers are very picky about formats they can > > handle although underlying formats can hold much more codecs > > AVI for example can pretty much hold any codec that have fourcc code. > > > > > > also note that sinks of muxers sometimes are named video_[0-9] and > > sometimes video_[0-9][0-9] > > > > while it is possible to avoid referencing those names directly, still it > > would be better to have standard naming > > > Unfortunately this can't be changed in the stable development series :/ Nether I need it, even gst-launch supports omitting the sources, but you say about stable version, so you will change this when you start next dev version? > > > > > - There are many pipeline stalls. > > it might be no joy to sit for a 4 hours to figure out > > 'where did I forgot to put that 'queue' > > > > Yes thats annoying, but imho its not easy to detect this from within gstreamer. > If you know a way, eternal fame awaits you. > > > > > also without magic 'videorate' that drops frames on demand (right?) it > > is very hard/impossible to create working pipeline, (not to mention that > > you absolutely need queues after each encoder) > > > I can't follow here. Many elements implement qos and they will take messures on > high cpu load like dropping frames. The queues are a nice concept for > multithreading. The ptoblem is that input seems to send extra frames there (its a mms stream), so without videorate, output stream still A/V synced, but audio skips a lot, videorate on the other hand ensures that video frames are skipped, and now A/V sync is perfect. > > > > also generally there is no way to copy a raw stream from one container > > to another. > Thats wrong. > > > > I for example tried to put asf stream (captured into file) ( gstream > > doesn't seem to support seeking in it) to avi, which is possible and is > > done nicely by mencoder. > Of course you can't put an asf into an avi. Both are containers. > Besides seeking in asf works, seeking on mms streams very recently got implemented. > > > > Gstreamer just stalls. > > Tell us what pipelines you have used and we might be able to help. gst-launch \ \ mmssrc location="mms://live1.wm.skynews.servecast.net/skynews_wmlz_live300k" ! queue ! asfdemux name=demuxer \ \ demuxer.audio_00 ! queue ! ffmux_asf name=muxer \ \ demuxer.video_00 ! queue ! muxer. \ \ muxer. ! queue ! fakesink -v also tried avimux, matroskamux, and both even don't have (again...) caps for wma/wmv. On the other hand mencoder does the same job perfectly. I first save the mms stream, and I tried to give saved stream to gstreamer, and got same results > > > > > I figured out that the problem was that intially input stream contained > > one big (~ 27 KB) chunk of video data, and then audio, but this made > > asfdemux stall, as it can't pull both video and audio data seperatly > > > > In other words demuxers can easily stall if input stream isn't evenly > > interleaved. > > Yes, use big enough queues after the demuxer, or even better use multiqueue. > > > > > > > Best regards, > > Maxim Levitsky > > > > PS: do you know whenever gstreamer can add an index to avi file? > > avimux could take a round of improvements. Not sure about that. > > long mail = lots of answers. lets look at the issues one by one > Stefan > > > > > I use this command to record a live asf stream: > > > > > >> #source description > >> export SOURCE_PROTOCOL=mmssrc > >> export SOURCE_DEMUXER=asfdemux > >> export SOURCE_AUDIO_CODEC=ffdec_wmav2 > >> export SOURCE_VIDEO_CODEC=ffdec_wmv3 > >> > >> #destanation description > >> export DEST_PROTOCOL=filesink > >> export DEST_MUXER=avimux > >> export DEST_AUDIO_CODEC=lame > >> export DEST_VIDEO_CODEC=xvidenc > > > >> gst-launch \ > >> $SOURCE_PROTOCOL location=$SOURCE_LOCATION ! queue ! $SOURCE_DEMUXER name=demuxer \ > >> \ > >> demuxer. ! queue ! $SOURCE_VIDEO_CODEC ! videorate ! postproc_default ! tee name=raw_video \ > >> demuxer. ! queue ! $SOURCE_AUDIO_CODEC ! tee name=raw_audio \ > >> \ > >> raw_video. ! queue leaky=1 ! $DEST_VIDEO_CODEC ! queue ! $DEST_MUXER name=muxer \ > >> raw_audio. ! queue leaky=1 ! $DEST_AUDIO_CODEC ! queue ! muxer. \ > >> \ > >> raw_video. ! queue ! gconfvideosink \ > >> raw_audio. ! queue ! gconfaudiosink \ > >> \ > >> muxer. ! $DEST_PROTOCOL location=$DEST_LOCATION \ > > > > It works, but resulting file can be seeking to non key-frames, which > > obliviously make seeking very unpleasant. > > (since muxers are limited to such small number of codecs, can't they > > know which frames are key and which not ?) > > > > > > Best regards, Maxim Levitsky From bilboed at gmail.com Fri Jan 30 12:37:00 2009 From: bilboed at gmail.com (Edward Hervey) Date: Fri, 30 Jan 2009 12:37:00 +0100 Subject: [gst-devel] gst-openmax repository is gone Message-ID: <1233315420.3853.29.camel@localhost> Hi all, The gst-openmax git repository is currently deactivated until Felipe Contreras gives us another repository. Edward From paul at linuxaudiosystems.com Fri Jan 30 13:04:50 2009 From: paul at linuxaudiosystems.com (Paul Davis) Date: Fri, 30 Jan 2009 13:04:50 +0100 Subject: [gst-devel] VST Audio Plug in approach using Gstreamer and Linux OS with ARM Cortex A8 processors In-Reply-To: References: Message-ID: <1233317090.2722.10.camel@dhin> > nfo/gstreamer-devel > Hello Paul, > > This Terry Shultz, > > I was wondering is we could have a chat by phone. I am based in Los Angeles > and I was hoping we could talk a bit more on the path in my earlier emails. > > My Cell# is 949.254.1159 Terry, sorry, I missed this (I only read gst-dev mail occasionally). I'm pretty busy these days and also in Berlin until Feb 17th, making me 9hrs offset from you, which I find fairly tricky. I find IRC to be a good substitute for the phone, but we could skype instead if we could find a good time slot. From brijesh.ksingh at gmail.com Fri Jan 30 14:40:17 2009 From: brijesh.ksingh at gmail.com (Brijesh Singh) Date: Fri, 30 Jan 2009 07:40:17 -0600 Subject: [gst-devel] Possible bug in v4l2src element - gst-plugins-good-0.10.10 In-Reply-To: <1233313436.23330.5.camel@mini.centricular.net> References: <1b712b840901291530l4021665ajfba708c7a140da51@mail.gmail.com> <1233313436.23330.5.camel@mini.centricular.net> Message-ID: <1b712b840901300540k572960b5r8918419710c95601@mail.gmail.com> Done Bug#569820. Thanks Brijesh Singh On Fri, Jan 30, 2009 at 5:03 AM, Tim-Philipp M?ller wrote: > On Thu, 2009-01-29 at 17:30 -0600, Brijesh Singh wrote: >> Hello, >> >> I was porting v4l2src element on TI Davinci platform and found a >> possible bug in v4l2src element. > > Could you file a bug in bugzilla please? > > http://gstreamer.freedesktop.org/bugs/ > > Cheers > -Tim > > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Confusion is always the most honest response. From vaisakh.n at nestgroup.net Fri Jan 30 14:43:36 2009 From: vaisakh.n at nestgroup.net (vaisakh.n) Date: Fri, 30 Jan 2009 19:13:36 +0530 Subject: [gst-devel] plugin Message-ID: <49830408.6080906@nestgroup.net> Hi, I written a codec plugin with the help of " plugin writers guide" in the gstreamer site .. when i tried to run that pipeline with ./gst-launch-0.10 an error comes like this ./gst-launch-0.10 filesrc location=test.mp3 ! examplefilter ! audioconvert ! osssink thanks in advance ERROR: from element /GstPipeline:pipeline0/GstAudioConvert:audioconvert0: not negotiated Additional debug info: gstbasetransform.c(1815): gst_base_transform_handle_buffer (): /GstPipeline:pipeline0/GstAudioConvert:audioconvert0: not negotiated ***** Confidentiality Statement/Disclaimer ***** This message and any attachments is intended for the sole use of the intended recipient. It may contain confidential information. Any unauthorized use, dissemination or modification is strictly prohibited. If you are not the intended recipient, please notify the sender immediately then delete it from all your systems, and do not copy, use or print. Internet communications are not secure and it is the responsibility of the recipient to make sure that it is virus/malicious code exempt. The company/sender cannot be responsible for any unauthorized alterations or modifications made to the contents. If you require any form of confirmation of the contents, please contact the company/sender. The company/sender is not liable for any errors or omissions in the content of this message. From pcarlson at kgisystems.com Fri Jan 30 16:05:06 2009 From: pcarlson at kgisystems.com (Peter Carlson) Date: Fri, 30 Jan 2009 07:05:06 -0800 Subject: [gst-devel] error compiling In-Reply-To: <1233311232.23330.1.camel@mini.centricular.net> References: <003001c9825c$d2c137d0$7843a770$@com> <1233311232.23330.1.camel@mini.centricular.net> Message-ID: <005801c982ec$22b79880$6826c980$@com> Problem was out of date m4 peter > -----Original Message----- > From: Tim-Philipp M?ller [mailto:t.i.m at zen.co.uk] > Sent: Friday, January 30, 2009 2:27 AM > To: gstreamer-devel at lists.sourceforge.net > Subject: Re: [gst-devel] error compiling > > On Thu, 2009-01-29 at 13:59 -0800, Peter Carlson wrote: > > > I am trying to compile gstreamer on redhat 9 (with mostly updated > libraries) > > I have tried both 0.10.11 and the most current version. > > In 0.10.11 I get an error > > /usr/local/share/bison/yacc.c:37: /usr/bin/m4: ERROR: Recursion limit > of 250 > > exceeded, use -L to change it > > /usr/local/bin/binson: subsidiary program `/usr/bin/mb' failed (exit > status > > 1) > > What are you trying to compile exactly? The most current version of > GStreamer (core/base) is 0.10.22. > > Cheers > -Tim > > > > ----------------------------------------------------------------------- > ------- > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From thiagossantos at gmail.com Fri Jan 30 16:13:16 2009 From: thiagossantos at gmail.com (thiagoss) Date: Fri, 30 Jan 2009 12:13:16 -0300 Subject: [gst-devel] Building failing for qtwrapper In-Reply-To: References: <497B3E34.6050604@hora-obscura.de> Message-ID: On Wed, Jan 28, 2009 at 1:47 PM, thiagoss wrote: > > > On Sat, Jan 24, 2009 at 2:13 PM, Stefan Kost wrote: > >> thiagoss schrieb: >> > Qtwraper building is failing (cvs head), I don't know if this is a >> > misconfiguration in my system or in Makefile, so I'm sending this mail. >> > The error output is: >> > >> > Making all in qtwrapper >> > cd ../.. && /bin/sh >> > /Users/thiagoss/Work/GStreamer/head/gst-plugins-bad/missing --run >> > automake-1.10 --gnu sys/qtwrapper/Makefile >> > configure.ac:111 : warning: >> > AC_CACHE_VAL(g_support_dowhile_macros, ...): suspicious cache-id, must >> > contain _cv_ to be cached >> > ../../lib/autoconf/general.m4:1974: AC_CACHE_VAL is expanded from... >> > ../../lib/autoconf/general.m4:1994: AC_CACHE_CHECK is expanded from... >> > common/m4/gst-dowhile.m4:8: AG_GST_CHECK_DOWHILE_MACROS is expanded >> from... >> > configure.ac:111 : the top level >> > configure.ac:143 : warning: >> > AC_CACHE_VAL(GST_FIONREAD_IN_SYS_IOCTL, ...): suspicious cache-id, must >> > contain _cv_ to be cached >> > m4/gst-fionread.m4:1: GST_CHECK_FIONREAD is expanded from... >> > configure.ac:143 : the top level >> > configure.ac:143 : warning: >> > AC_CACHE_VAL(GST_FIONREAD_IN_SYS_FILIO, ...): suspicious cache-id, must >> > contain _cv_ to be cached >> > cd ../.. && /bin/sh ./config.status sys/qtwrapper/Makefile depfiles >> > config.status: creating sys/qtwrapper/Makefile >> > config.status: executing depfiles commands >> > /bin/sh ../../libtool --tag=CC --tag=disable-static --mode=compile gcc >> > -DHAVE_CONFIG_H -I. -I../.. -I../../gst-libs -I../../gst-libs >> > -D_REENTRANT -I/usr/local/include/gstreamer-0.10 >> > -I/opt/local/include/glib-2.0 -I/opt/local/lib/glib-2.0/include >> > -I/opt/local/include -I/opt/local/include/libxml2 -Wall >> > -Wdeclaration-after-statement -Werror -fno-strict-aliasing -g >> > -D_REENTRANT -I/usr/local/include/gstreamer-0.10 >> > -I/opt/local/include/glib-2.0 -I/opt/local/lib/glib-2.0/include >> > -I/opt/local/include -I/opt/local/include/libxml2 -g -O2 -MT >> > libgstqtwrapper_la-qtwrapper.lo -MD -MP -MF >> > .deps/libgstqtwrapper_la-qtwrapper.Tpo -c -o >> > libgstqtwrapper_la-qtwrapper.lo `test -f 'qtwrapper.c' || echo >> > './'`qtwrapper.c >> > libtool: compile: gcc -DHAVE_CONFIG_H -I. -I../.. -I../../gst-libs >> > -I../../gst-libs -D_REENTRANT -I/usr/local/include/gstreamer-0.10 >> > -I/opt/local/include/glib-2.0 -I/opt/local/lib/glib-2.0/include >> > -I/opt/local/include -I/opt/local/include/libxml2 -Wall >> > -Wdeclaration-after-statement -Werror -fno-strict-aliasing -g >> > -D_REENTRANT -I/usr/local/include/gstreamer-0.10 >> > -I/opt/local/include/glib-2.0 -I/opt/local/lib/glib-2.0/include >> > -I/opt/local/include -I/opt/local/include/libxml2 -g -O2 -MT >> > libgstqtwrapper_la-qtwrapper.lo -MD -MP -MF >> > .deps/libgstqtwrapper_la-qtwrapper.Tpo -c qtwrapper.c -fno-common -DPIC >> > -o .libs/libgstqtwrapper_la-qtwrapper.o >> > In file included from qtwrapper.c:49: >> > qtwrapper.h:53:20: error: Movies.h: No such file or directory >> > qtwrapper.c: In function 'plugin_init': >> > qtwrapper.c:58: error: 'OSErr' undeclared (first use in this function) >> > qtwrapper.c:58: error: (Each undeclared identifier is reported only once >> > qtwrapper.c:58: error: for each function it appears in.) >> > qtwrapper.c:58: error: syntax error before 'status' >> > qtwrapper.c:69: error: 'status' undeclared (first use in this function) >> > cc1: warnings being treated as errors >> > qtwrapper.c:69: warning: implicit declaration of function 'EnterMovies' >> > make[3]: *** [libgstqtwrapper_la-qtwrapper.lo] Error 1 >> > make[2]: *** [all-recursive] Error 1 >> > make[1]: *** [all-recursive] Error 1 >> > make: *** [all] Error 2 >> > >> > How to fix this? >> >> It seems that configure belives you have all the needed external libs and >> headers, but in fact you don't. Do you have a Movies.h? Maybe you can look >> at >> config.log and check what its looking for. >> >> Stefan >> > > # find / -name "Movies.h" > /Developer/Headers/FlatCarbon/Movies.h > /Developer/SDKs/MacOSX10.4u.sdk/Developer/Headers/CFMCarbon/Movies.h > > /Developer/SDKs/MacOSX10.4u.sdk/Developer/Headers/CFMCarbon/QuickTime/Movies.h > /Developer/SDKs/MacOSX10.4u.sdk/Developer/Headers/FlatCarbon/Movies.h > > /Developer/SDKs/MacOSX10.4u.sdk/System/Library/Frameworks/QuickTime.framework/Versions/A/Headers/Movies.h > /Developer/SDKs/MacOSX10.5.sdk/Developer/Headers/FlatCarbon/Movies.h > > /Developer/SDKs/MacOSX10.5.sdk/System/Library/Frameworks/QuickTime.framework/Versions/A/Headers/Movies.h > /System/Library/Frameworks/QuickTime.framework/Versions/A/Headers/Movies.h > > > Maybe it is looking in the wrong places. > > The config.log has this: > > configure:30272: *** checking feature: QuickTime wrapper *** > configure:30276: *** for plug-ins: qtwrapper *** > configure:30324: checking QuickTime/Movies.h usability > configure:30341: gcc -c -g -O2 conftest.c >&5 > configure:30348: $? = 0 > configure:30362: result: yes > configure:30366: checking QuickTime/Movies.h presence > configure:30381: gcc -E conftest.c > configure:30388: $? = 0 > configure:30402: result: yes > configure:30435: checking for QuickTime/Movies.h > configure:30442: result: yes > configure:30464: *** These plugins will be built: qtwrapper > > Found the problem, the test scripts at configure look for QuickTime/file.h, the plugin files were trying to use Quicktime/file.h, that matters in a case sensitive system. It is now fixed at the repository. > > >> >> > >> > -- >> > Thiago Sousa Santos >> > >> > Embedded Systems and Pervasive Computing Lab (Embedded) >> > Center of Electrical Engineering and Informatics (CEEI) >> > Federal University of Campina Grande (UFCG) >> > >> > >> > ------------------------------------------------------------------------ >> > >> > >> ------------------------------------------------------------------------------ >> > This SF.net email is sponsored by: >> > SourcForge Community >> > SourceForge wants to tell your story. >> > http://p.sf.net/sfu/sf-spreadtheword >> > >> > >> > ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > gstreamer-devel mailing list >> > gstreamer-devel at lists.sourceforge.net >> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> ------------------------------------------------------------------------------ >> This SF.net email is sponsored by: >> SourcForge Community >> SourceForge wants to tell your story. >> http://p.sf.net/sfu/sf-spreadtheword >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > > > -- > Thiago Sousa Santos > > > Embedded Systems and Pervasive Computing Lab (Embedded) > Center of Electrical Engineering and Informatics (CEEI) > Federal University of Campina Grande (UFCG) > -- Thiago Sousa Santos Embedded Systems and Pervasive Computing Lab (Embedded) Center of Electrical Engineering and Informatics (CEEI) Federal University of Campina Grande (UFCG) -------------- next part -------------- An HTML attachment was scrubbed... URL: From msmith at xiph.org Fri Jan 30 18:03:47 2009 From: msmith at xiph.org (Michael Smith) Date: Fri, 30 Jan 2009 09:03:47 -0800 Subject: [gst-devel] plugin In-Reply-To: <49830408.6080906@nestgroup.net> References: <49830408.6080906@nestgroup.net> Message-ID: <3c1737210901300903s33ef588bmb370b1cac0030d5@mail.gmail.com> On Fri, Jan 30, 2009 at 5:43 AM, vaisakh.n wrote: > Hi, > I written a codec plugin with the help of " plugin writers guide" in > the gstreamer site .. > > when i tried to run that pipeline with ./gst-launch-0.10 > an error comes like this > ./gst-launch-0.10 filesrc location=test.mp3 ! examplefilter ! > audioconvert ! osssink > > thanks in advance > > ERROR: from element > /GstPipeline:pipeline0/GstAudioConvert:audioconvert0: not negotiated > Additional debug info: > gstbasetransform.c(1815): gst_base_transform_handle_buffer (): > /GstPipeline:pipeline0/GstAudioConvert:audioconvert0: > not negotiated Well, all this tells us is that there's a bug in your element. You haven't given us any details to help you with - it's probably something to do with how you set caps on your buffers. Mike From marcsabat at gmail.com Fri Jan 30 20:37:49 2009 From: marcsabat at gmail.com (Marc) Date: Fri, 30 Jan 2009 20:37:49 +0100 Subject: [gst-devel] Start recording from a webcam already playing Message-ID: <8fc9ead30901301137x2b86dca4h2397b4ead932d74a@mail.gmail.com> Hi, I am trying to build an application more or less like Cheese ( http://projects.gnome.org/cheese/). However I would like to switch from just showing the webcam to a recording state "smoothly" when the user press the record button. So far I have tried the following approaches without succes: 1. Changing location property of filesink while playing (from /dev/null to a real path). This is not possible (filesink tells me so). 2. Open 2 different pipelines both of them pointing to the same v4l2src (no success since only one pipeline can access the webcam at once). Since these approaches were not working I tried to set the pipeline to STATE_NULL, change the location of the filesink, and then set the pipeline to STATE_PLAYING again. However, when setting the pipeline to STATE_NULL, no STATE_CHANGE event is thrown. Is this the expected behavior ? Any ideas ? Can the smooth "play-to-record" be accomplished with GStreamer ? Thank you PD. For the record: I am programming in Python and the pipeline I am trying to manage is the following: gst-launch-0.10 v4l2src ! tee name=videoout ! queue ! videorate ! video/x-raw-yuv,fps=15 ! queue ! theoraenc quality=60 ! queue ! muxout. pulsesrc ! audio/x-raw-int,rate=22000,channels=1,width=16 ! queue ! audioconvert ! vorbisenc ! queue ! muxout. oggmux name=muxout ! filesink location=test.ogg videoout. ! queue ! ffmpegcolorspace ! ximagesink -- Marc S. -------------- next part -------------- An HTML attachment was scrubbed... URL: From thaytan at noraisin.net Sat Jan 31 01:29:48 2009 From: thaytan at noraisin.net (Jan Schmidt) Date: Sat, 31 Jan 2009 00:29:48 +0000 Subject: [gst-devel] element specific bus messages In-Reply-To: <4981D1FD.2000007@hora-obscura.de> References: <1232467772.6039.40.camel@calculus> <3c1737210901201016p35764100v985520ff16b47d3d@mail.gmail.com> <1232480165.6039.61.camel@calculus> <3c1737210901281013h4c137762qe7fd9ae4af19f722@mail.gmail.com> <4981D1FD.2000007@hora-obscura.de> Message-ID: <1233361788.31252.10.camel@fancy-ubuntu> On Thu, 2009-01-29 at 17:57 +0200, Stefan Kost wrote: > > > > They are uploaded wrongly. It needs a gtkdoc-rebase --online and then > uploading. > > Stefan > Hmmm. How does that look in terms of actual commands to execute? Can we make the gtkdoc-rebase execution part of the upload make target? J. -- Jan Schmidt From felipe.contreras at gmail.com Sat Jan 31 10:37:53 2009 From: felipe.contreras at gmail.com (Felipe Contreras) Date: Sat, 31 Jan 2009 11:37:53 +0200 Subject: [gst-devel] [gst-embedded] Writing a Video Sink and using hardware decoders. In-Reply-To: References: <1233311104.3853.28.camel@localhost> Message-ID: <94a0d4530901310137m7f34c9cck2e681edebb1e78b8@mail.gmail.com> On Fri, Jan 30, 2009 at 12:45 PM, Daniel James Laird wrote: > Many thanks for the reply, > we are having similar discussions internally as well. > > My question back would be > > Could I do the following? > filesrc -> demux -> My plugin (Codec/Decoder) -> FakeSink > > This would mean I am more like the standard model. However the Fakesink does very little just satisfies the standard gstreamer model. But it means that the closed video pipe (that's handles avsync etc, hides all datapointers etc is all left as it is). > I just feed my Codec/Decoder plugin with encoded video (write to the device (+ metadata like timestamps)) and everytime I decode a frame I output something to keep the FakeSink (effectively /dev/null) happy. If you do that you'll loose A/V sync. Whatever sync you choose on the pipeline must receive proper timestamps. It's OK to write a dec+sink elements, that's what we use in Maemo, but they need to act as sinks (inherit from basesink) so that they handle pre-rolling and A/V sink, etc. -- Felipe Contreras From felipe.contreras at gmail.com Sat Jan 31 11:39:50 2009 From: felipe.contreras at gmail.com (Felipe Contreras) Date: Sat, 31 Jan 2009 12:39:50 +0200 Subject: [gst-devel] gst-openmax repository is gone In-Reply-To: <1233315420.3853.29.camel@localhost> References: <1233315420.3853.29.camel@localhost> Message-ID: <94a0d4530901310239s2d8687a4ue551d7721b314208@mail.gmail.com> On Fri, Jan 30, 2009 at 1:37 PM, Edward Hervey wrote: > Hi all, > > The gst-openmax git repository is currently deactivated until Felipe > Contreras gives us another repository. It's now back but for some reason the cgit UI isn't showing it: Here's the git url: git://anongit.freedesktop.org/gstreamer/gst-openmax -- Felipe Contreras From ivan.murashko at gmail.com Sat Jan 31 12:18:38 2009 From: ivan.murashko at gmail.com (Ivan Murashko) Date: Sat, 31 Jan 2009 14:18:38 +0300 Subject: [gst-devel] Streaming FLV files in http In-Reply-To: <497353E1.9060409@gmail.com> References: <497353E1.9060409@gmail.com> Message-ID: <9c6ce7150901310318j4800c04al12ac2ca5b152f375@mail.gmail.com> Hello Pierpaolo, I have the same problem (want to stream with gstreamer via http). After a small investigation I found that it's impossible to do it now with available plugins. In my opinion it can be done only by creating an own plugin. The good start point is the tcpclient sink plugin (from base set). Do you want to cooperate with it? May be my info is obsolete and you've found what you need? -- Best regards, Ivan On Sun, Jan 18, 2009 at 7:08 PM, Pierpaolo Gull?a wrote: > Hi guys , > > this is my first messagge on the Gstreamer's mailing list. > > SO i try to explain my need. > > I want to develop an application that take and. mpg files (encoded with > ffmpeg , with mpeg2video for the video and mp2 for the audio). > > I want one files , and encoding to the flv on the fly and streaming in > http this flv > > I see that with gstreamer it's possible , but I dont' know the command > line instructions that I must to use on my Ubuntu 8.0.4 Box. > > Can you provide to me some links or some instructions? > > Best Regards. > > Goodbye > > Pierpaolo > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From kotyo at zamba.fm Sat Jan 31 14:31:47 2009 From: kotyo at zamba.fm (Tamas Korodi) Date: Sat, 31 Jan 2009 14:31:47 +0100 Subject: [gst-devel] Changing v4l2src's capabilities without stopping the pipeline Message-ID: Hi Guys, I would like to write an application, which captures video from a webcam, with v4l2src, and displays it with an xvimagesink. I use the next pipeline v4l2src -> capsfilter -> xvimagesink. My webcam can produce 320x240 and 640x480 images, with capsfilter I request 320x240 first. Later I would like to change the resolution to 640x480 on the fly. I tried to reach this by the following steps: - Block the v4l2src's src pad - Unlink capsfilter from v4l2src and xvimagesink - Set capsfilter's state to NULL - Create a new capsfilter with 640x480 resolution - Link the new capsfilter to xvimagesink, and link v4l2src to the new capsfilter - Set the new capsfilter's state to PAUSED - Unblock v4l2src's src pad. it doesn't work for me. How can I solve this problem? Thanks for the answers! BR, Tamas -------------- next part -------------- An HTML attachment was scrubbed... URL: From felipe.contreras at gmail.com Sat Jan 31 14:32:26 2009 From: felipe.contreras at gmail.com (Felipe Contreras) Date: Sat, 31 Jan 2009 15:32:26 +0200 Subject: [gst-devel] GIT conversion ... DONE In-Reply-To: <1232581228.7969.34.camel@joder> References: <1232518526.7969.12.camel@joder> <1232531364.4111.1.camel@fancy-ubuntu> <1232581228.7969.34.camel@joder> Message-ID: <94a0d4530901310532x44872af6rb34f03fae8ae8d35@mail.gmail.com> On Thu, Jan 22, 2009 at 1:40 AM, Edward Hervey wrote: > On Wed, 2009-01-21 at 09:49 +0000, Jan Schmidt wrote: >> On Wed, 2009-01-21 at 07:15 +0100, Edward Hervey wrote: >> > Hello hackers, >> > >> > All the main GStreamer modules are now converted to git and up. >> > >> > You can view them here http://cgit.freedesktop.org/gstreamer/ >> > The url to access each module is at the bottom of each of those >> > modules. >> > >> >> Hey hey, >> >> Thanks for getting the conversion done and everything online. Jonathon >> pointed to this commit as problematic: >> >> http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=a3e95ebfc9586534fa26d7ea59f34987f6e78ad1 >> >> The author attribution is incorrect. I guess where there's one, there's >> others too. How would we go about fixing that now? Is it worth fixing? >> >> J. > > I don't think this is a blocker issue. The authorname is still valid > in the commit message, and those are *historical* commits. > > After discussing in #gstreamer, we can add graft points later on if we > actually see actual problematic issues/commits without modifying the > head commit sha1 (and therefore not screwing up everybody > cloning/branching off the repositories). I just found a couple of UTF-8 issues: http://cgit.freedesktop.org/gstreamer/gst-ffmpeg/commit/?id=3627dbca5a51aeaced656455f43ff6c26a7ca6d1 -- Felipe Contreras From apolatynski at gmail.com Sat Jan 31 14:48:57 2009 From: apolatynski at gmail.com (=?UTF-8?Q?Andrzej_Polaty=C5=84ski?=) Date: Sat, 31 Jan 2009 14:48:57 +0100 Subject: [gst-devel] Video surveillance application Message-ID: Hello, I'm writing a simple video surveillance application. I started with creating simple GUI and basic gst pipeline for viewing live video stream. Now I want to add some video analyzing code to perform abandoned object detection. For this I need to grab video frames. I found this element gdkpixbufsink and thought that it can be used for my task, but I couldn't force it to work yet. As of now, my pipeline looks like this: v4l2src -> ffmpegcolorspace -> tee -> xvimagesink \-> gdkpixbufsink Is my approach good? Does the gdkpixbufsink need a separate thread? Is there any other way of getting video frames? My second problem is how to draw over output video images, so that user could see for example an red outline indicating a detected object through xvimagesink. I see that there are some elements that can overlay text, but I can't find any element that could overlay an image. -- Thanks, Andrzej Polaty?ski From sebastian.droege at collabora.co.uk Sat Jan 31 15:09:35 2009 From: sebastian.droege at collabora.co.uk (Sebastian =?ISO-8859-1?Q?Dr=F6ge?=) Date: Sat, 31 Jan 2009 15:09:35 +0100 Subject: [gst-devel] Streaming FLV files in http In-Reply-To: <9c6ce7150901310318j4800c04al12ac2ca5b152f375@mail.gmail.com> References: <497353E1.9060409@gmail.com> <9c6ce7150901310318j4800c04al12ac2ca5b152f375@mail.gmail.com> Message-ID: <1233410975.9696.4.camel@odin.lan> Am Samstag, den 31.01.2009, 14:18 +0300 schrieb Ivan Murashko: > Hello Pierpaolo, > > I have the same problem (want to stream with gstreamer via http). > After a small investigation I found that it's impossible to do it now > with available plugins. In my opinion it can be done only by creating > an own plugin. > > The good start point is the tcpclient sink plugin (from base set). Do > you want to cooperate with it? May be my info is obsolete and you've > found what you need? You should use the flvdemux plugin from gst-plugins-bad. It supports streaming over HTTP and other stuff ;) -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: Dies ist ein digital signierter Nachrichtenteil URL: From t.i.m at zen.co.uk Sat Jan 31 15:27:21 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Sat, 31 Jan 2009 14:27:21 +0000 Subject: [gst-devel] Git commit messages Message-ID: <1233412041.29516.23.camel@mini.centricular.net> Hi all, maybe I'm alone with this, but I think it'd be nice if our commit message summary lines were a bit more self-explanatory. In particular, I think it'd be nice if we could start prefixing the summary lines with the plugin/element/baseclass in question, e.g. playbin: fix this and that blademux: .. basesrc: ... controller: ... or the like. This would make it easier to see from the commit messages / summary lines whether a commit is 'interesting' or not, and to separate signal from noise, and to generally follow what's going on. (Yes, I'm aware that the changed files are listed both in gitk/giggle and in the commit mail.) In the same vein, please don't use words like 'here' in a summary line ('Don't do this here' is not a great summary), and make it clear whether code was changed or not (e.g. 'Fix typo' could mean code or docs). Flame away. Cheers -Tim From felipe.contreras at gmail.com Sat Jan 31 16:04:12 2009 From: felipe.contreras at gmail.com (Felipe Contreras) Date: Sat, 31 Jan 2009 17:04:12 +0200 Subject: [gst-devel] Git commit messages In-Reply-To: <1233412041.29516.23.camel@mini.centricular.net> References: <1233412041.29516.23.camel@mini.centricular.net> Message-ID: <94a0d4530901310704ra580a57v3b96ccee22e14fd3@mail.gmail.com> On Sat, Jan 31, 2009 at 4:27 PM, Tim-Philipp M?ller wrote: > Hi all, > > maybe I'm alone with this, but I think it'd be nice if our commit > message summary lines were a bit more self-explanatory. > > In particular, I think it'd be nice if we could start prefixing the > summary lines with the plugin/element/baseclass in question, e.g. > > playbin: fix this and that > blademux: .. > basesrc: ... > controller: ... > > or the like. This would make it easier to see from the commit messages / > summary lines whether a commit is 'interesting' or not, and to separate > signal from noise, and to generally follow what's going on. (Yes, I'm > aware that the changed files are listed both in gitk/giggle and in the > commit mail.) > > In the same vein, please don't use words like 'here' in a summary line > ('Don't do this here' is not a great summary), and make it clear whether > code was changed or not (e.g. 'Fix typo' could mean code or docs). +1 >From git documentation: Though not required, it's a good idea to begin the commit message with a single short (less than 50 character) line summarizing the change, followed by a blank line and then a more thorough description. Tools that turn commits into email, for example, use the first line on the Subject: line and the rest of the commit in the body. Also, many other tools benefit from a short single line summary (gitk, cgit, etc.) Besides, it would be great if we could have some basic guideline: 6 months from now, would I be able to understand what the commit is doing and why by just reading the message? -- Felipe Contreras From arnout at mind.be Sat Jan 31 17:16:49 2009 From: arnout at mind.be (Arnout Vandecappelle) Date: Sat, 31 Jan 2009 17:16:49 +0100 Subject: [gst-devel] Start recording from a webcam already playing In-Reply-To: <8fc9ead30901301137x2b86dca4h2397b4ead932d74a@mail.gmail.com> References: <8fc9ead30901301137x2b86dca4h2397b4ead932d74a@mail.gmail.com> Message-ID: <200901311716.49441.arnout@mind.be> On Friday 30 January 2009 20:37:49 Marc wrote: > I am trying to build an application more or less like Cheese > (http://projects.gnome.org/cheese/). However I would like to switch from > just showing the webcam to a recording state "smoothly" when the user press > the record button. > > So far I have tried the following approaches without succes: > Changing location property of filesink while playing (from /dev/null to a > real path). This is not possible (filesink tells me so). Open 2 different > pipelines both of them pointing to the same v4l2src (no success since only > one pipeline can access the webcam at once). Since these approaches were > not working I tried to set the pipeline to STATE_NULL, change the location > of the filesink, and then set the pipeline to STATE_PLAYING again. However, > when setting the pipeline to STATE_NULL, no STATE_CHANGE event is thrown. > Is this the expected behavior ? > > Any ideas ? Can the smooth "play-to-record" be accomplished with GStreamer > ? > > Thank you > PD. For the record: I am programming in Python and the pipeline I am trying > to manage is the following: > > gst-launch-0.10 v4l2src ! tee name=videoout ! queue ! videorate ! > video/x-raw-yuv,fps=15 ! queue ! theoraenc quality=60 ! queue ! muxout. > pulsesrc ! audio/x-raw-int,rate=22000,channels=1,width=16 ! queue ! > audioconvert ! vorbisenc ! queue ! muxout. oggmux name=muxout ! filesink > location=test.ogg videoout. ! queue ! ffmpegcolorspace ! ximagesink You can add the videoout tail of the pipeline dynamically. Start without it, and when the user presses the record button, add it to the pipeline and link it (and set each element to PLAYING individually). It's not even necessary to pause the pipeline. It's more difficult to smoothly stop the recording, though. You'll have to block the queue element behind the tee (see docs/design/part-block.txt in the source tree). In addition, you have to push an EOS into it to make sure the muxer writes duration and index to the file. A last caveat is the timestamps. The buffer timestamps start running as soon as the v4l2src starts playing, which makes it appear in the output file as if it starts with a big hole. The only way I found to fix that is to add an adapter element that shifts the buffer timestamps (see http://bugzilla.gnome.org/show_bug.cgi?id=561224 ). Regard, Arnout -- Arnout Vandecappelle arnout at mind be Senior Embedded Software Architect +32-16-286540 Essensium/Mind http://www.mind.be G.Geenslaan 9, 3001 Leuven, Belgium BE 872 984 063 RPR Leuven LinkedIn profile: http://www.linkedin.com/in/arnoutvandecappelle GPG fingerprint: D206 D44B 5155 DF98 550D 3F2A 2213 88AA A1C7 C933 From thaytan at noraisin.net Sat Jan 31 17:22:06 2009 From: thaytan at noraisin.net (Jan Schmidt) Date: Sat, 31 Jan 2009 16:22:06 +0000 Subject: [gst-devel] gst-openmax repository is gone In-Reply-To: <94a0d4530901310239s2d8687a4ue551d7721b314208@mail.gmail.com> References: <1233315420.3853.29.camel@localhost> <94a0d4530901310239s2d8687a4ue551d7721b314208@mail.gmail.com> Message-ID: <1233418926.31252.11.camel@fancy-ubuntu> On Sat, 2009-01-31 at 12:39 +0200, Felipe Contreras wrote: > On Fri, Jan 30, 2009 at 1:37 PM, Edward Hervey wrote: > > Hi all, > > > > The gst-openmax git repository is currently deactivated until Felipe > > Contreras gives us another repository. > > It's now back but for some reason the cgit UI isn't showing it: > > Here's the git url: > git://anongit.freedesktop.org/gstreamer/gst-openmax > It's there now. cgit takes a little while to notice new repos. J. -- Jan Schmidt From ylatuya at gmail.com Sat Jan 31 18:41:31 2009 From: ylatuya at gmail.com (Andoni Morales) Date: Sat, 31 Jan 2009 18:41:31 +0100 Subject: [gst-devel] Start recording from a webcam already playing In-Reply-To: <200901311716.49441.arnout@mind.be> References: <8fc9ead30901301137x2b86dca4h2397b4ead932d74a@mail.gmail.com> <200901311716.49441.arnout@mind.be> Message-ID: <772db3280901310941wea985e1h14c4d2aeb672b4c4@mail.gmail.com> > > > It's more difficult to smoothly stop the recording, though. You'll have > to > block the queue element behind the tee (see docs/design/part-block.txt in > the > source tree). In addition, you have to push an EOS into it to make sure > the > muxer writes duration and index to the file. > > A last caveat is the timestamps. The buffer timestamps start running as > soon > as the v4l2src starts playing, which makes it appear in the output file as > if > it starts with a big hole. The only way I found to fix that is to add an > adapter element that shifts the buffer timestamps (see > http://bugzilla.gnome.org/show_bug.cgi?id=561224 ). The 'reset' property must be setted to true just before dinamically linking the encoder tail or after linking it? Is it possible to record various segments in the same file using this method? I mean, for the first segment you set the base_time to 0, but if I want to encode another segment in the same file, by unliking the encoding tail and relinking it later, if I set the 'reset' property to TRUE this will reset the base_time to 0 again, when it should be setted to the elapsed time of the previous segment. One last thing, if I pause the recording, by unlinking the encoding tail and setting its state to PAUSED, how do I send an EOS event to the muxer (which is not linked anymore) if I defenitively want to stop the recording? -------------- next part -------------- An HTML attachment was scrubbed... URL: From felipe.contreras at gmail.com Sat Jan 31 18:48:12 2009 From: felipe.contreras at gmail.com (Felipe Contreras) Date: Sat, 31 Jan 2009 19:48:12 +0200 Subject: [gst-devel] gst-openmax repository is gone In-Reply-To: <1233418926.31252.11.camel@fancy-ubuntu> References: <1233315420.3853.29.camel@localhost> <94a0d4530901310239s2d8687a4ue551d7721b314208@mail.gmail.com> <1233418926.31252.11.camel@fancy-ubuntu> Message-ID: <94a0d4530901310948n1d18be63hf1f2f228a4193be@mail.gmail.com> On Sat, Jan 31, 2009 at 6:22 PM, Jan Schmidt wrote: > On Sat, 2009-01-31 at 12:39 +0200, Felipe Contreras wrote: >> On Fri, Jan 30, 2009 at 1:37 PM, Edward Hervey wrote: >> > Hi all, >> > >> > The gst-openmax git repository is currently deactivated until Felipe >> > Contreras gives us another repository. >> >> It's now back but for some reason the cgit UI isn't showing it: >> >> Here's the git url: >> git://anongit.freedesktop.org/gstreamer/gst-openmax >> > > It's there now. cgit takes a little while to notice new repos. Where? It still doesn't seem to work at: http://cgit.freedesktop.org/gstreamer/gst-openmax/ -- Felipe Contreras From airteknology at gmail.com Sat Jan 31 19:59:08 2009 From: airteknology at gmail.com (=?UTF-8?B?UGllcnBhb2xvIEd1bGzCiGE=?=) Date: Sat, 31 Jan 2009 18:59:08 -0000 Subject: [gst-devel] Streaming FLV files in http In-Reply-To: <1233410975.9696.4.camel@odin.lan> References: <497353E1.9060409@gmail.com> <9c6ce7150901310318j4800c04al12ac2ca5b152f375@mail.gmail.com> <1233410975.9696.4.camel@odin.lan> Message-ID: <49849F5F.7000801@gmail.com> Hi Sebastain , thanks for your help! Now i do another search in google , and I found this blog http://blog.mycroes.nl/2008_06_01_archive.html Now I have more clear how gstreamer works. I must to try to take a mpeg2 files , create by ffmpeg , installed on the sistem , and streaming this using the flvdemux plugin from gst-plugins-bad Do You have any idea about the command that we must to use ? Thanks for the help Best regards Pierpaolo Sebastian ? ha scritto: > Am Samstag, den 31.01.2009, 14:18 +0300 schrieb Ivan Murashko: > >> Hello Pierpaolo, >> >> I have the same problem (want to stream with gstreamer via http). >> After a small investigation I found that it's impossible to do it now >> with available plugins. In my opinion it can be done only by creating >> an own plugin. >> >> The good start point is the tcpclient sink plugin (from base set). Do >> you want to cooperate with it? May be my info is obsolete and you've >> found what you need? >> > > You should use the flvdemux plugin from gst-plugins-bad. It supports > streaming over HTTP and other stuff ;) >