[gst-devel] gstreamer-devel Digest, Vol 25, Issue 14
Rahul Nikose
rahul.nikose at gmail.com
Wed Jun 11 09:23:46 CEST 2008
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... :)
On Wed, Jun 11, 2008 at 12:15 PM, <
gstreamer-devel-request at lists.sourceforge.net> wrote:
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> 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 <tristan at sat.qc.ca>
> Subject: Re: [gst-devel] problem with 8 channel interleaved audio over
> rtp
> To: Michael Smith <msmith at xiph.org>
> Cc: "gstreamer-devel at lists.sourceforge.net"
> <gstreamer-devel at lists.sourceforge.net>
> Message-ID: <484EBCC6.4020800 at sat.qc.ca>
> 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" <b-liu at ti.com>
> Subject: Re: [gst-devel] oss audio pipeline doesn't work when debug on
> To: <gstreamer-devel at lists.sourceforge.net>
> Message-ID:
> <74DB28182CB793438AC65D689C58BBF8015D2460 at dlee10.ent.ti.com>
> 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: gstreamer-devel-bounces at lists.sourceforge.net
> [mailto:gstreamer-devel-bounces at lists.sourceforge.net] On Behalf Of Liu,
> Bin
> Sent: Tuesday, June 10, 2008 9:28 AM
> To: gstreamer-devel at lists.sourceforge.net
> 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:ds at schleef.org]
> Sent: Monday, June 09, 2008 8:13 PM
> To: Liu, Bin
> Cc: gstreamer-devel at lists.sourceforge.net
> 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...
>
>
> ------------------------------------------------------------------------
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>
>
> ------------------------------
>
> Message: 3
> Date: Tue, 10 Jun 2008 23:05:59 +0100
> From: Jan Schmidt <thaytan at noraisin.net>
> Subject: [gst-devel] Core/Base/Python 2nd pre-releases tomorrow.
> To: gstreamer-devel at lists.sourceforge.net
> Message-ID: <1213135559.6118.5.camel at 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 (http://bugzilla.gnome.org)
> >
> >
> http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.19.2.tar.bz2
> >
> >
> http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.19.2.tar.bz2
> > and
> >
> http://gstreamer.freedesktop.org/src/gst-python/pre/gst-python-0.10.11.2.tar.bz2
> >
> > Cheers,
> > Jan
> --
> Jan Schmidt <thaytan at noraisin.net>
>
>
>
>
> ------------------------------
>
> Message: 4
> Date: Tue, 10 Jun 2008 19:50:39 -0400
> From: Rov Juvano <rovjuvano at users.sourceforge.net>
> Subject: Re: [gst-devel] Making Scaletempo good
> To: gstreamer-devel at lists.sourceforge.net
> Message-ID: <20080610195039.nomail at owht.52.off>
>
> On Mon, Jun 09, 2008 at 11:41:10AM +0200, Sebastian Dröge 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:
> > > http://sourceforge.net/project/showfiles.php?group_id=220192
> > >
> > > Git repos at:
> > > http://repo.or.cz/w/gst-scaletempo-rj.git
> > > http://repo.or.cz/w/gst-scaletempo-demo-rj.git
> >
> > 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 <nitinm76 at yahoo.com>
> Subject: Re: [gst-devel] MP3, AAC and MPEG Audio codecs
> To: Michael Smith <msmith at xiph.org>
> Cc: gstreamer-devel at lists.sourceforge.net
> Message-ID: <527213.37422.qm at web50101.mail.re2.yahoo.com>
> Content-Type: text/plain; charset=utf-8
>
> Hello Micahel!
>
> --- On Tue, 10/6/08, Michael Smith <msmith at xiph.org> wrote:
>
> > From: Michael Smith <msmith at xiph.org>
> > Subject: Re: [gst-devel] MP3, AAC and MPEG Audio codecs
> > To: nitinm76 at yahoo.com
> > Cc: gstreamer-devel at lists.sourceforge.net
> > Date: Tuesday, 10 June, 2008, 10:24 PM
> > On Tue, Jun 10, 2008 at 3:53 AM, Nitin Mahajan
> > <nitinm76 at yahoo.com> 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 http://uk.messenger.yahoo.com
>
>
>
> ------------------------------
>
> Message: 6
> Date: Wed, 11 Jun 2008 11:53:36 +0530
> From: Ramana Reddy Polaka <Ramana_Polaka at infosys.com>
> Subject: [gst-devel] AAC stream play with faad gst plugin :
> negotiation problem
> To: "gstreamer-devel at lists.sourceforge.net"
> <gstreamer-devel at lists.sourceforge.net>
> Message-ID:
> <
> BC658D4967D21E4AAF3C12EFB33B5AC114C4344316 at BLRKECMBX01.ad.infosys.com>
>
> 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
>
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> End of gstreamer-devel Digest, Vol 25, Issue 14
> ***********************************************
>
--
Thanks and warm Regards
Rahul S. Nikose
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