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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, < [hidden email]> 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 <[hidden email]>
Subject: Re: [gst-devel] problem with 8 channel interleaved audio over
rtp
To: Michael Smith <[hidden email]>
Cc: "[hidden email]"
<[hidden email]>
Message-ID: <[hidden email]>
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" <[hidden email]>
Subject: Re: [gst-devel] oss audio pipeline doesn't work when debug on
To: <[hidden email]>
Message-ID:
<[hidden email]>
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: [hidden email]
[mailto:[hidden email]] On Behalf Of Liu,
Bin
Sent: Tuesday, June 10, 2008 9:28 AM
To: [hidden email]
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:[hidden email]]
Sent: Monday, June 09, 2008 8:13 PM
To: Liu, Bin
Cc: [hidden email]
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 <[hidden email]>
Subject: [gst-devel] Core/Base/Python 2nd pre-releases tomorrow.
To: [hidden email]
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 (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 <[hidden email]>
------------------------------
Message: 4
Date: Tue, 10 Jun 2008 19:50:39 -0400
From: Rov Juvano <[hidden email]>
Subject: Re: [gst-devel] Making Scaletempo good
To: [hidden email]
Message-ID: <[hidden email]>
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 <[hidden email]>
Subject: Re: [gst-devel] MP3, AAC and MPEG Audio codecs
To: Michael Smith <[hidden email]>
Cc: [hidden email]
Message-ID: <[hidden email]>
Content-Type: text/plain; charset=utf-8
Hello Micahel!
--- On Tue, 10/6/08, Michael Smith <[hidden email]> wrote:
> From: Michael Smith <[hidden email]>
> Subject: Re: [gst-devel] MP3, AAC and MPEG Audio codecs
> To: [hidden email]
> Cc: [hidden email]
> Date: Tuesday, 10 June, 2008, 10:24 PM
> On Tue, Jun 10, 2008 at 3:53 AM, Nitin Mahajan
> <[hidden email]> 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 <[hidden email]>
Subject: [gst-devel] AAC stream play with faad gst plugin :
negotiation problem
To: "[hidden email]"
<[hidden email]>
Message-ID:
<[hidden email]>
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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