hi everybody,          I m trying to develop demuxer which will open file and parse it into audio and video.           I use gst-template tool for creating one simple filter plugin (filter with one sink and one source ...it will forward incoming buffer as it is ) . I used sub code gstplugin during creation of this filter. Now i want to build element which itself act as filesrc + demuxer .  i.e             _______            |            |--------> Video            |            |            |_______|--------->Audio This element will use API for opening file for parsing file (mp4 ) and will spilt file into video and audio sample which will forwarded to h264 decoder and AMR decoder resp.                 Now I have build one gstremer element which wrapp this API's and used them for it internal purpose (i.e opening and parsing).          Please guid me through various step required to build this element...from my exp. in bulding simple filter element ....i used gstplugin stub code and get help Plugin development manual for understanding various code snippets.....but now I don't have any guild line or stubcode to follow...                     So request you friend to state what are the step I must follow to get job done. How to proceed...  Thank You  ......Have nice day...   :) <div class="gmail_quote"> On Wed, Jun 11, 2008 at 12:15 PM, <<a href="mailto:gstreamer-devel-request@lists.sourceforge.net" target="_blank">gstreamer-devel-request@lists.sourceforge.net</a>> wrote: <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"> Send gstreamer-devel mailing list submissions to        <a href="mailto:gstreamer-devel@lists.sourceforge.net" target="_blank">gstreamer-devel@lists.sourceforge.net</a> To subscribe or unsubscribe via the World Wide Web, visit        <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a> or, via email, send a message with subject or body 'help' to        <a href="mailto:gstreamer-devel-request@lists.sourceforge.net" target="_blank">gstreamer-devel-request@lists.sourceforge.net</a> You can reach the person managing the list at        <a href="mailto:gstreamer-devel-owner@lists.sourceforge.net" target="_blank">gstreamer-devel-owner@lists.sourceforge.net</a> When replying, please edit your Subject line so it is more specific than "Re: Contents of gstreamer-devel digest..." Today's Topics:   1. Re: problem with 8 channel interleaved audio over rtp      (Tristan Matthews)   2. Re: oss audio pipeline doesn't work when debug on (Liu, Bin)   3. Core/Base/Python 2nd pre-releases tomorrow. (Jan Schmidt)   4. Re: Making Scaletempo good (Rov Juvano)   5. Re: MP3, AAC and MPEG Audio codecs (Nitin Mahajan)   6. AAC stream play with faad gst plugin : negotiation problem      (Ramana Reddy Polaka) ---------------------------------------------------------------------- Message: 1 Date: Tue, 10 Jun 2008 13:41:26 -0400 From: Tristan Matthews <<a href="mailto:tristan@sat.qc.ca" target="_blank">tristan@sat.qc.ca</a>> Subject: Re: [gst-devel] problem with 8 channel interleaved audio over        rtp To: Michael Smith <<a href="mailto:msmith@xiph.org" target="_blank">msmith@xiph.org</a>> Cc: "<a href="mailto:gstreamer-devel@lists.sourceforge.net" target="_blank">gstreamer-devel@lists.sourceforge.net</a>"        <<a href="mailto:gstreamer-devel@lists.sourceforge.net" target="_blank">gstreamer-devel@lists.sourceforge.net</a>> Message-ID: <<a href="mailto:484EBCC6.4020800@sat.qc.ca" target="_blank">484EBCC6.4020800@sat.qc.ca</a>> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi Mike, Michael Smith wrote: > This is probably (roughly) the right approach, but there are some > problems. You want float audio (not int) - that's what vorbisenc > requires. You want to set 'channel-positions' (not > 'channel-position'). And I'm not sure if you can use channel positions > from gst-launch, you might need to write an actual application. > You're right about the launch line not working for channel-positions, fortunately this was going into a C app anyway. I ended up fixing the issue by setting the channel-positions argument of interleave to the 8 channel layout specified in gst-plugins-base/ext/vorbis/vorbisenc.c (other layouts may work as well, I'm not sure), kind of like the 2 channel example from gst-plugins-bad/tests/check/elements/interleave.c Thanks for your help, and thanks also to slomo for some earlier feedback on this issue. -Tristan ------------------------------ Message: 2 Date: Tue, 10 Jun 2008 16:08:40 -0500 From: "Liu, Bin" <<a href="mailto:b-liu@ti.com" target="_blank">b-liu@ti.com</a>> Subject: Re: [gst-devel] oss audio pipeline doesn't work when debug on To: <<a href="mailto:gstreamer-devel@lists.sourceforge.net" target="_blank">gstreamer-devel@lists.sourceforge.net</a>> Message-ID:        <<a href="mailto:74DB28182CB793438AC65D689C58BBF8015D2460@dlee10.ent.ti.com" target="_blank">74DB28182CB793438AC65D689C58BBF8015D2460@dlee10.ent.ti.com</a>> Content-Type: text/plain;       charset="us-ascii" The issue got resolved. The video codec accidentally decreased the oss driver DMA priority. -Bin. -----Original Message----- From: <a href="mailto:gstreamer-devel-bounces@lists.sourceforge.net" target="_blank">gstreamer-devel-bounces@lists.sourceforge.net</a> [mailto:<a href="mailto:gstreamer-devel-bounces@lists.sourceforge.net" target="_blank">gstreamer-devel-bounces@lists.sourceforge.net</a>] On Behalf Of Liu, Bin Sent: Tuesday, June 10, 2008 9:28 AM To: <a href="mailto:gstreamer-devel@lists.sourceforge.net" target="_blank">gstreamer-devel@lists.sourceforge.net</a> Subject: Re: [gst-devel] oss audio pipeline doesn't work when debug on Thanks, Dave, The issue I face is the pipeline can play either the audio or video stream, but not both together. (The mpeg4 decoder runs on a hardware similar as a DSP.) The pipeline seems running but no video and audio outputs. After a while when I Ctrl+C to stop the pipeline, few video frames will show on TV. The pipeline I use is gst-launch -v --gst-debug-level=2 \        filesrc location=t.avi ! avidemux name=t  t.audio_00 ! \        queue ! mad ! osssink  t.video_00 ! \        queue ! gdecoder Codec=1 ! fbvideosink Is there any way to give the audio thread higher priority to avoid dropping samples? Please let me know if a debug log will give a better idea what is going on. Thanks, -Bin. -----Original Message----- From: David Schleef [mailto:<a href="mailto:ds@schleef.org" target="_blank">ds@schleef.org</a>] Sent: Monday, June 09, 2008 8:13 PM To: Liu, Bin Cc: <a href="mailto:gstreamer-devel@lists.sourceforge.net" target="_blank">gstreamer-devel@lists.sourceforge.net</a> Subject: Re: [gst-devel] oss audio pipeline doesn't work when debug on On Mon, Jun 09, 2008 at 03:35:13PM -0500, Liu, Bin wrote: > Hi, > > I am new to gstreamer. I cross-compiled gst to my ARM board. I have no > problem to play the mp3 audio stream from a movie clip using the > following pipeline: > >       gst-launch -v --gst-debug-level=2 \ >       filesrc location=t.avi ! avidemux ! queue ! mad ! osssink > > But if I turn on the debug level higher than level 2 for any component, > I cannot hear anything from the speakers, for example using the > following pipeline: This is not surprising.  Audio is automatically dropped when the buffers arrive at the sink late.  Debugging produces a lot of output, and if the output device is slow or has a small buffer, it will likely fill up and cause all GStreamer processing to stop.  On a desktop system, output to an xterm is both fast and has a large buffer, so you're unlikely to ever see the problem.  The best way to solve this on an embedded system is using ssh to log into the system and/or pipe the debug output out to a file. It is also possible to write a custom debug handler that manages the output in a specific way.  For one client, I created a custom debug handler that emulated a simple web server, and connecting to this web server gave a constant stream of gstreamer debug output in a web browser. dave... ------------------------------------------------------------------------ - Check out the new SourceForge.net Marketplace. It's the best place to buy or sell services for just about anything Open Source. <a href="http://sourceforge.net/services/buy/index.php" target="_blank">http://sourceforge.net/services/buy/index.php</a> _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.sourceforge.net" target="_blank">gstreamer-devel@lists.sourceforge.net</a> <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a> ------------------------------ Message: 3 Date: Tue, 10 Jun 2008 23:05:59 +0100 From: Jan Schmidt <<a href="mailto:thaytan@noraisin.net" target="_blank">thaytan@noraisin.net</a>> Subject: [gst-devel] Core/Base/Python 2nd pre-releases tomorrow. To: <a href="mailto:gstreamer-devel@lists.sourceforge.net" target="_blank">gstreamer-devel@lists.sourceforge.net</a> Message-ID: <1213135559.6118.5.camel@fancy-ubuntu> Content-Type: text/plain Hi all, I'm going to be cutting the 2nd pre-release tarballs of Core/Base/Python tomorrow afternoon sometime (Dublin time). It'd be great if people could flag out patches for inclusion for me so we can make sure they go in. Cheers, Jan. On Thu, 2008-06-05 at 01:01 +0100, Jan Schmidt wrote: > Hi all, > > The Core, Base and Python modules are frozen for 0.10.20, 0.10.20 and > 0.10.12 releases respectively. I've just uploaded the first pre-release > tarballs. Please test them and file any found issues in bugzilla as > usual (<a href="http://bugzilla.gnome.org" target="_blank">http://bugzilla.gnome.org</a>) > > <a href="http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.19.2.tar.bz2" target="_blank">http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.19.2.tar.bz2</a> > > <a href="http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.19.2.tar.bz2" target="_blank">http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.19.2.tar.bz2</a> > and > <a href="http://gstreamer.freedesktop.org/src/gst-python/pre/gst-python-0.10.11.2.tar.bz2" target="_blank">http://gstreamer.freedesktop.org/src/gst-python/pre/gst-python-0.10.11.2.tar.bz2</a> > > Cheers, > Jan -- Jan Schmidt <<a href="mailto:thaytan@noraisin.net" target="_blank">thaytan@noraisin.net</a>> ------------------------------ Message: 4 Date: Tue, 10 Jun 2008 19:50:39 -0400 From: Rov Juvano <<a href="mailto:rovjuvano@users.sourceforge.net" target="_blank">rovjuvano@users.sourceforge.net</a>> Subject: Re: [gst-devel] Making Scaletempo good To: <a href="mailto:gstreamer-devel@lists.sourceforge.net" target="_blank">gstreamer-devel@lists.sourceforge.net</a> Message-ID: <20080610195039.nomail@owht.52.off> On Mon, Jun 09, 2008 at 11:41:10AM +0200, Sebastian Dr�ge wrote: > Am Donnerstag, den 05.06.2008, 17:23 -0400 schrieb Rov Juvano: > > I'm working on a trick-mode plugin that maintains the audio > > pitch when playback rate != 1.0.  Similar to soundtouch/pitch > > but in C and with no external dependencies and interactive. > > > > I have it in a working state, but need help making it good > > enough for GStreamer. > > > > I've also put together a demo app.  I'm not too concerned > > with robustness of the demo, but if someone can figure out > > why, after awhile, my GtkEntry's stop updating and my menus > > get garbled, that would be much appreciated. > > > > You can grab the code at: > > <a href="http://sourceforge.net/project/showfiles.php?group_id=220192" target="_blank">http://sourceforge.net/project/showfiles.php?group_id=220192</a> > > > > Git repos at: > > <a href="http://repo.or.cz/w/gst-scaletempo-rj.git" target="_blank">http://repo.or.cz/w/gst-scaletempo-rj.git</a> > > <a href="http://repo.or.cz/w/gst-scaletempo-demo-rj.git" target="_blank">http://repo.or.cz/w/gst-scaletempo-demo-rj.git</a> > > I tested it a bit and it sounds good... would you be interested to have > your code added to gst-plugins-bad? If so it would be nice if you could > file a bug for this with a patch for adding it to gst-plugins-bad and > I'll care for it :) Is that the normal process?  File a bug and have it added to -bad.  I'm willing to work to make it -good, but I'm unsure of the standards and procedures. Bug #537700.  Do you need me to make a patch against -bad? -- rovjuvano ------------------------------ Message: 5 Date: Tue, 10 Jun 2008 20:16:02 -0700 (PDT) From: Nitin Mahajan <<a href="mailto:nitinm76@yahoo.com" target="_blank">nitinm76@yahoo.com</a>> Subject: Re: [gst-devel] MP3, AAC and MPEG Audio codecs To: Michael Smith <<a href="mailto:msmith@xiph.org" target="_blank">msmith@xiph.org</a>> Cc: <a href="mailto:gstreamer-devel@lists.sourceforge.net" target="_blank">gstreamer-devel@lists.sourceforge.net</a> Message-ID: <<a href="mailto:527213.37422.qm@web50101.mail.re2.yahoo.com" target="_blank">527213.37422.qm@web50101.mail.re2.yahoo.com</a>> Content-Type: text/plain; charset=utf-8 Hello Micahel! --- On Tue, 10/6/08, Michael Smith <<a href="mailto:msmith@xiph.org" target="_blank">msmith@xiph.org</a>> wrote: > From: Michael Smith <<a href="mailto:msmith@xiph.org" target="_blank">msmith@xiph.org</a>> > Subject: Re: [gst-devel] MP3, AAC and MPEG Audio codecs > To: <a href="mailto:nitinm76@yahoo.com" target="_blank">nitinm76@yahoo.com</a> > Cc: <a href="mailto:gstreamer-devel@lists.sourceforge.net" target="_blank">gstreamer-devel@lists.sourceforge.net</a> > Date: Tuesday, 10 June, 2008, 10:24 PM > On Tue, Jun 10, 2008 at 3:53 AM, Nitin Mahajan > <<a href="mailto:nitinm76@yahoo.com" target="_blank">nitinm76@yahoo.com</a>> wrote: > > HI! > > > > I tried to play MP3, AAC media files with gstreamer > with gst-ffmpeg plugin. I could not play both of them. > > FFmpeg doesn't have an aac decoder, and the mp3 decoder > is very bad. > GStreamer won't autoplug the mp3 decoder, though you > can use it if you > build a pipeline manually. Thanks for the exaplaination. > > There's a good quality mp3 decoder in gst-plugins-ugly > ('mad'), and an > aac decoder in gst-plugins-bad ('faad'). > I would test them, but would you recommend gst-plugins-bad from Quality perspective? > > > > Whether MP3 and AAC have been removed from ffmpeg in > gst-ffmpeg? If, yes whether it has been removed from > configuration or through source code? > > > > >From Quality and legal perspectives which ones > would be the right set of plugins for MP3, AAC and MPEG > Audio for playback purpose? > > From a legal perspective, you'll want to get a licensed > decoder > plugin. Fluendo has a free mp3 decoder plugin (fully > licensed). I will try Fluendo aslo, but whats your input on this Quality wise?  You'll > need to pay for any legal AAC decoder. If I am paying the license, quality wise which one is a good AAC decoder plugin for gstreamer? Any decoder plugin can be created in gstreamer? Thanks and regards -Nitin Send instant messages to your online friends <a href="http://uk.messenger.yahoo.com" target="_blank">http://uk.messenger.yahoo.com</a> ------------------------------ Message: 6 Date: Wed, 11 Jun 2008 11:53:36 +0530 From: Ramana Reddy Polaka <<a href="mailto:Ramana_Polaka@infosys.com" target="_blank">Ramana_Polaka@infosys.com</a>> Subject: [gst-devel] AAC stream play with faad gst plugin :        negotiation problem To: "<a href="mailto:gstreamer-devel@lists.sourceforge.net" target="_blank">gstreamer-devel@lists.sourceforge.net</a>"        <<a href="mailto:gstreamer-devel@lists.sourceforge.net" target="_blank">gstreamer-devel@lists.sourceforge.net</a>> Message-ID:        <<a href="mailto:BC658D4967D21E4AAF3C12EFB33B5AC114C4344316@BLRKECMBX01.ad.infosys.com" target="_blank">BC658D4967D21E4AAF3C12EFB33B5AC114C4344316@BLRKECMBX01.ad.infosys.com</a>> Content-Type: text/plain; charset="us-ascii" Hi, I tried to play an aac stream using faad bad plugin. $ gst-typefind AAC_ADTS_LC_24_193_3.aac  AAC_ADTS_LC_24_193_3.aac - audio/mpeg, framed=(boolean)false, mpegversion=(int)4 Command used: ---------------------- gst-launch filesrc location=AAC_ADTS_LC_24_193_3.aac  ! faad ! alsasink Error log ------------ Setting pipeline to PAUSED ... Pipeline is PREROLLING ... ERROR: from element /pipeline0/filesrc0: Internal data flow error. Additional debug info: gstbasesrc.c(2165): gst_base_src_loop (): /pipeline0/filesrc0: streaming task paused, reason not-negotiated (-4) ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... FREEING pipeline ... It looks there is problem in negotiation. Should I use any parser? I tried using ffdemux_mov_mp4_m4a_3gp_3g2_mj2. But no use. Any suggestions pls? Regards, Ramana **************** CAUTION - Disclaimer ***************** This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify the sender by e-mail and delete the original message. Further, you are not to copy, disclose, or distribute this e-mail or its contents to any other person and any such actions are unlawful. This e-mail may contain viruses. 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It's the best place to buy or sell services for just about anything Open Source. <a href="http://sourceforge.net/services/buy/index.php" target="_blank">http://sourceforge.net/services/buy/index.php</a> ------------------------------ _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.sourceforge.net" target="_blank">gstreamer-devel@lists.sourceforge.net</a> <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel" target="_blank">https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a> End of gstreamer-devel Digest, Vol 25, Issue 14 *********************************************** </blockquote></div> -- Thanks and warm Regards Rahul S. Nikose