Re: gstreamer-devel Digest, Vol 43, Issue 49

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Re: gstreamer-devel Digest, Vol 43, Issue 49

sreerenj


On Thu, Dec 31, 2009 at 1:16 PM, <[hidden email]> wrote:
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Today's Topics:

  1. Re: gstreamer-devel Digest, Vol 43, Issue 47 (sreerenj b)


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Message: 1
Date: Thu, 31 Dec 2009 13:15:51 +0530
From: sreerenj b <[hidden email]>
Subject: Re: [gst-devel] gstreamer-devel Digest, Vol 43, Issue 47
To: [hidden email]
Message-ID:
       <[hidden email]>
Content-Type: text/plain; charset="iso-8859-1"

On Wed, Dec 30, 2009 at 11:36 PM, <
[hidden email]> wrote:

> Send gstreamer-devel mailing list submissions to
>        [hidden email]
>
> 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
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>
> You can reach the person managing the list at
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>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of gstreamer-devel digest..."
>
>
> Today's Topics:
>
>   1. trouble with playbin2 and audio track (kitone)
>   2. Re: level element in playbin2 (Stefan Kost)
>   3. Re: gstreamer clock (Stefan Kost)
>   4. Re: gstreamer in a non-multimedia area? (Stefan Kost)
>   5. H.264 decoder sink problem ([hidden email])
>   6. Re: gstreamer in a non-multimedia area? (Leonardo Sandoval)
>   7. Re: Playbin2 and Audio-CDs (regomodo)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Wed, 30 Dec 2009 13:18:54 +0100
> From: kitone <[hidden email]>
> Subject: [gst-devel] trouble with playbin2 and audio track
> To: [hidden email]
> Message-ID: <[hidden email]>
> Content-Type: text/plain; charset=US-ASCII
>
> Hi everyone,
>
> I'm using playbin2 to play video and I've add the ability to change
> audio track in the player. This doesn't work very well. When I switch
> to an other track there's little problem like no sound, need to seek to
> have sound again or some little freeze.
>
> So there's a best way to change track ?
>
> Do I need to define the state of the pipeline to PAUSED, change the
> audio track and replay again ? Or can I do this on the fly ?
>
> code :
> void Player::set_current_audio(int track) {
>  g_object_set (playbin, "current-audio", track, NULL);
> }
>
> Best regards,
> kitone
>
>
>
> ------------------------------
>
> Message: 2
> Date: Wed, 30 Dec 2009 16:30:15 +0200
> From: Stefan Kost <[hidden email]>
> Subject: Re: [gst-devel] level element in playbin2
> To: Discussion of the development of GStreamer
>        <[hidden email]>
> Message-ID: <[hidden email]>
> Content-Type: text/plain; charset=ISO-8859-1
>
> hi,
>
> Am 28.12.2009 14:34, schrieb J. Rios:
> > Hello,
> >    Im tryign to incorporate the level element into a playbin2. This is
> > the significant part of what I have tried
> >
> >     GstElement * outputsink;
> >     outputsink = gst_element_factory_make( "gconfaudiosink", "audio-sink"
> );
> > ...
> >     GstElement * m_Playbin = gst_element_factory_make( "playbin2", "play"
> );
> > ...
> >     GstElement * level = gst_element_factory_make( "level", "levelctrl"
> );
> >
> >     GstElement * replay = gst_element_factory_make( "rgvolume",
> > "replaygain" );
> >
> >
> >     gst_bin_add( GST_BIN( m_Playbin ), level );
> >     gst_bin_add( GST_BIN( m_Playbin ), replay );
> >     g_object_set( G_OBJECT( m_Playbin ), "audio-sink", outputsink, NULL
> );
> >
> >     g_object_set( level, "message", true, NULL );
> >
> >     g_object_set( G_OBJECT( m_Playbin ), "flags", 0x02|0x10, NULL );
> >
> >     gst_bus_add_watch( gst_pipeline_get_bus( GST_PIPELINE( m_Playbin ) ),
> >             ( GstBusFunc ) gst_bus_async_callback, this );
> >
> >
> this cannot work as such. The elements need to be linked in the right
> order.
> What you need to do, is to create an own bin, add level, replay and the
> audiosink there, link them and set your bin as the audiosink. You need to
> add a
> ghostpad to the bin and set it to the first (yet unlinked) sink pad of you
> elements.
>
> Stefan
>
>
> > The problem I have is that I dont get the Element messages. All the rest
> > works fine including the replay-gain volumen but I dont get the level
> > messages in the callback function.
> > What I want to do is detect when the sound level is below an user
> > defined level so I can skip to next track.
> >
> > I dunno if this is the correct place to put it but I have been tryign to
> > search for this with no success. Please if u know what is wrong or how
> > it should work let me know
> >
> > Thanks in advance
> > J.Rios
> >
> >
> >
> >
> ------------------------------------------------------------------------------
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>
>
>
> ------------------------------
>
> Message: 3
> Date: Wed, 30 Dec 2009 16:33:26 +0200
> From: Stefan Kost <[hidden email]>
> Subject: Re: [gst-devel] gstreamer clock
> To: Discussion of the development of GStreamer
>        <[hidden email]>
> Cc: sreerenj b <[hidden email]>
> Message-ID: <[hidden email]>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Am 30.12.2009 08:19, schrieb sreerenj b:
> >
> > HI,
> >
> > Is there any way to get some signal or tick from the pipeline depending
> > upon the data transferd?(For example, suppose if i change the streaming
> > server playback rate to twice the actual rate,then i should needs to get
> > some identification from pipeline in each second or in  some periodic
> > manner.)
>
> You could use gst_clock_new_periodic_id() together with a
> gst_clock_id_wait() or
> gst_clock_id_wait_async()
>
> Stefan
>
> > --
> > Sreerenj B
> > [hidden email] <mailto:[hidden email]>
> > mob: +91 9739469496
> >
> >
>

 

  Hi ,

   I captured the GstClockID of the pipeline in streaming-server

 id=gst_clock_new_periodic_id(media->media->pipeline->clock,start_time,interval);

And  then setted up the asynchronous  callback using the following.

gst_clock_id_wait_async(id,(GstClockCallback *)TickCallback,"Tick sending");

but  evenif i "stopped" or "paused" the client side player , i am getting the callback " TickCallback" in the server side on specified intervals!!!!!!! I take the clock of the pipeline currently running, but why i am getting this call back even i stopped the pipeline?Is there any way to rectify this?

I setted up these settings in   gst-rtsp-server/gst/rtsp-server/rtsp-client.c ...inside the handle_play() function.


sreerenj



> >
> >
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> >
> >
> > _______________________________________________
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>
>
>
>
> ------------------------------
>
> Message: 4
> Date: Wed, 30 Dec 2009 16:36:37 +0200
> From: Stefan Kost <[hidden email]>
> Subject: Re: [gst-devel] gstreamer in a non-multimedia area?
> To: Discussion of the development of GStreamer
>        <[hidden email]>
> Cc: Leonardo Sandoval <[hidden email]>
> Message-ID: <[hidden email]>
> Content-Type: text/plain; charset=ISO-8859-1
>
> hi,
>
> Am 29.12.2009 18:18, schrieb Leonardo Sandoval:
> > Hi,
> >
> > In the past, I worked in a project which consists of joining together
> several numerical codes (codes which solve some particular physic problem,
> like stars gravity or star mass evolution) in a single one, but this turned
> to be a very difficult problem due to different language/technologies.
> >
> > Recently, I moved to the multimedia field using gstreamer and seems  to
> be an excellent and natural 'glue' to put several things together under a
> single framework. Before I can dig deeper on this, I have a simple question:
> in the case of galaxy dynamics, we are not talking about frames but
> 'snapshots', where each snapshot contains the information of each star
> inside a galaxy (mass, position, velocity, gas composition, etc.), i.e. the
> state  of the galaxy at some time.
> >
> > I know that gstreamer does not care about the kind of data flowing, I
> just want to know if gstreamer has been used in other fields  rather than
> multimedia and if yes, which has been the experienced with this technology.
> >
>
> I am quite sure it can be done. WHat I wonder is how gstreamer might help
> you.
> For multimedia it gives you data flow synchronisation and format
> negotiation
> (and other things). I am not sure you would need those. What would be left
> is
> the plugin framework and some developments support.
>
> >
> > Happy new year!
> >
> Same to you.
>
> Stefan
> >
> >
> >       Encuentra las mejores recetas en Yahoo! Cocina.
> > http://mx.mujer.yahoo.com/cocina/
> >
> >
> ------------------------------------------------------------------------------
> > This SF.Net email is sponsored by the Verizon Developer Community
> > Take advantage of Verizon's best-in-class app development support
> > A streamlined, 14 day to market process makes app distribution fast and
> easy
> > Join now and get one step closer to millions of Verizon customers
> > http://p.sf.net/sfu/verizon-dev2dev
> > _______________________________________________
> > gstreamer-devel mailing list
> > [hidden email]
> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>
>
>
>
> ------------------------------
>
> Message: 5
> Date: Wed, 30 Dec 2009 20:15:13 +0530
> From: <[hidden email]>
> Subject: [gst-devel] H.264 decoder sink problem
> To: <[hidden email]>
> Message-ID:
>        <[hidden email]>
> Content-Type: text/plain; charset="us-ascii"
>
> Hi ,
>        I am developing H264 decoder plugin. In my current pipeline i
> took filesrc, myplugin and autovideosink elements.
> i was able to decode the .H264 input file but i am not able to view the
> video on sink side. The decoded data by writing into a file and able to
> run with yuv tool. but through sink i am not able to. Could anyone
> suggest what output buffer should i create and which capabilities need i
> to set?
>
>    Please its urgent...
>
> Thanks & Regards
> Mohan Reddy
>
> -------------- next part --------------
> An HTML attachment was scrubbed...
>
> ------------------------------
>
> Message: 6
> Date: Wed, 30 Dec 2009 09:14:38 -0800 (PST)
> From: Leonardo Sandoval <[hidden email]>
> Subject: Re: [gst-devel] gstreamer in a non-multimedia area?
> To: Discussion of the development of GStreamer
>        <[hidden email]>,        Stefan Kost
>        <[hidden email]>
> Message-ID: <[hidden email]>
> Content-Type: text/plain; charset=iso-8859-1
>
> I think may be gstreamer a good option at least for those two reasons you
> mentioned:
>
> * Syncronisation: for example, in a galaxy evolution, several physics
> (gravitation, star evolution, etc.) are occurring at the same time, so we
> need syncronisation to control  these computations.
>
> * format negotiation: currently, different codes outputs different data
> formats so most of the time the output of one program has to be converted in
> order to be the input of another program. Using standard formats specified
> in the pads/caps, these can be avoided, I believe.
>
> However, there is one problem. In a multimedia, we have all the data to be
> reproduced (encoded,decoded,etc) but in this case, starting from an initial
> snapshot (initial conditions) an new snapshot is produced (the snapshot at
> time t_0 + delta), then the latter is used to create the snapshot t_0 +
> 2*delta, etc, and I believe in gstreamer it is not possible that the output
> of a right element is the input of a left element, like in this diagram:
>
>         ________        ________        _______
>        |        |      |        |      |       |
> --|->---|        |----- |        |------|       |--->---|
>  |     |________|      |________|      |_______|       |
>  |                                                     |
>  |-----------------------------------------------------|
> Ing. Leonardo SandovalCel:  (044)  81 10 62 33 59Casa: (0181) 17 33 04 73
>
>
> --- El mi? 30-dic-09, Stefan Kost <[hidden email]> escribi?:
>
> > De:: Stefan Kost <[hidden email]>
> > Asunto: Re: [gst-devel] gstreamer in a non-multimedia area?
> > A: "Discussion of the development of GStreamer" <
> [hidden email]>
> > Cc: "Leonardo Sandoval" <[hidden email]>
> > Fecha: mi?rcoles 30 de diciembre de 2009, 8:36
> > hi,
> >
> > Am 29.12.2009 18:18, schrieb Leonardo Sandoval:
> > > Hi,
> > >
> > > In the past, I worked in a project which consists of
> > joining together several numerical codes (codes which solve
> > some particular physic problem, like stars gravity or star
> > mass evolution) in a single one, but this turned to be a
> > very difficult problem due to different
> > language/technologies.
> > >
> > > Recently, I moved to the multimedia field using
> > gstreamer and seems? to be an excellent and natural
> > 'glue' to put several things together under a single
> > framework. Before I can dig deeper on this, I have a simple
> > question: in the case of galaxy dynamics, we are not talking
> > about frames but 'snapshots', where each snapshot contains
> > the information of each star inside a galaxy (mass,
> > position, velocity, gas composition, etc.), i.e. the
> > state? of the galaxy at some time.?
> > >
> > > I know that gstreamer does not care about the kind of
> > data flowing, I just want to know if gstreamer has been used
> > in other fields? rather than multimedia and if yes,
> > which has been the experienced with this technology.
> > >
> >
> > I am quite sure it can be done. WHat I wonder is how
> > gstreamer might help you.
> > For multimedia it gives you data flow synchronisation and
> > format negotiation
> > (and other things). I am not sure you would need those.
> > What would be left is
> > the plugin framework and some developments support.
> >
> > >
> > > Happy new year!
> > >
> > Same to you.
> >
> > Stefan
> > >
> > >
> > >? ? ???Encuentra las mejores
> > recetas en Yahoo! Cocina.? ? ? ? ?
> > ? ? ? ? ? ???
> > > http://mx.mujer.yahoo.com/cocina/
> > >
> > >
> >
> ------------------------------------------------------------------------------
> > > This SF.Net email is sponsored by the Verizon
> > Developer Community
> > > Take advantage of Verizon's best-in-class app
> > development support
> > > A streamlined, 14 day to market process makes app
> > distribution fast and easy
> > > Join now and get one step closer to millions of
> > Verizon customers
> > > http://p.sf.net/sfu/verizon-dev2dev
> > > _______________________________________________
> > > gstreamer-devel mailing list
> > > [hidden email]
> > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
> >
> >
>
>
>      Encuentra las mejores recetas en Yahoo! Cocina.
> http://mx.mujer.yahoo.com/cocina/
>
>
>
> ------------------------------
>
> Message: 7
> Date: Wed, 30 Dec 2009 18:06:24 +0000
> From: regomodo <[hidden email]>
> Subject: Re: [gst-devel] Playbin2 and Audio-CDs
> To: Discussion of the development of GStreamer
>        <[hidden email]>
> Message-ID: <[hidden email]>
> Content-Type: Text/Plain;  charset="iso-8859-1"
>
> On Monday 28 December 2009 16:01:06 Tim-Philipp M?ller wrote:
> > On Mon, 2009-12-21 at 20:55 +0000, reg wrote:
> > > I'm trying to use playbin2 with audiocds. I can get a track to
> > > play,pause and seek fine but the moment I try and change the uri
> > > playbin2 just plays up a lot. These are the errors I get:
> > >
> > > (Gereqi.py:5019): GStreamer-WARNING **: Name queue is not unique in bin
> > > uridecodebin0, not adding
> > >
> > > (Gereqi.py:5019): GStreamer-CRITICAL **: gst_caps_get_structure:
> > > assertion `index < caps->structs->len' failed
> > >
> > > (Gereqi.py:5019): GStreamer-CRITICAL **: gst_structure_get_name:
> > > assertion `structure != NULL' failed
> > >
> > >
> > > However, I can use playbin2 again without restarting my app. I just
> > > can't use the audio-cd until I restart my app.
> > >
> > > Is it possible to use playbin2 with audio-cd's yet?
> >
> > This was a bug, it's fixed in git now and will be in the upcoming -base
> > 0.10.26 release.
> >
> > See https://bugzilla.gnome.org/show_bug.cgi?id=605219
> >
> > Cheers
> >  -Tim
> >
> >
> >
> >
> ---------------------------------------------------------------------------
> > --- This SF.Net email is sponsored by the Verizon Developer Community
> > Take advantage of Verizon's best-in-class app development support
> > A streamlined, 14 day to market process makes app distribution fast and
> >  easy Join now and get one step closer to millions of Verizon customers
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> > _______________________________________________
> > gstreamer-devel mailing list
> > [hidden email]
> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
> >
> I finally got around to trying out git. Uninstalled everything gst related
> on
> my system and built gstreamer,gst-plugins-base and gst-python (in that
> order)
> only adding --prefix=/usr/ to ./configure.
>
> However, I get an error when I try to play a cd track:
>
> 'Error: Your GStreamer installation is missing a plug-in.',
> 'gstplaysink.c(1483): gen_audio_chain ():
> /GstPlayBin2:player/GstPlaySink:playsink0'
>
> I see this is from gst-plugins-base but I cannot see an option to
> enable/disable this.
>
>
>
> ------------------------------
>
>
> ------------------------------------------------------------------------------
> This SF.Net email is sponsored by the Verizon Developer Community
> Take advantage of Verizon's best-in-class app development support
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> Join now and get one step closer to millions of Verizon customers
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> ------------------------------
>
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
>
>
> End of gstreamer-devel Digest, Vol 43, Issue 47
> ***********************************************
>



--
Sreerenj B
Software engineer,Carinov Networks Pvt Ltd
[hidden email]
mob: +91 9739469496
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End of gstreamer-devel Digest, Vol 43, Issue 49
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--
Sreerenj B
Software engineer,Carinov Networks Pvt Ltd
[hidden email]
mob: +91 9739469496

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Re: gstreamer-devel Digest, Vol 43, Issue 49

Stefan Sauer
Am 31.12.2009 21:22, schrieb sreerenj b:

>     > ------------------------------
>     >
>     > Message: 3
>     > Date: Wed, 30 Dec 2009 16:33:26 +0200
>     > From: Stefan Kost <[hidden email]
>     <mailto:[hidden email]>>
>     > Subject: Re: [gst-devel] gstreamer clock
>     > To: Discussion of the development of GStreamer
>     >        <[hidden email]
>     <mailto:[hidden email]>>
>     > Cc: sreerenj b <[hidden email] <mailto:[hidden email]>>
>     > Message-ID: <[hidden email]
>     <mailto:[hidden email]>>
>     > Content-Type: text/plain; charset=ISO-8859-1
>     >
>     > Am 30.12.2009 08:19, schrieb sreerenj b:
>     > >
>     > > HI,
>     > >
>     > > Is there any way to get some signal or tick from the pipeline
>     depending
>     > > upon the data transferd?(For example, suppose if i change the
>     streaming
>     > > server playback rate to twice the actual rate,then i should
>     needs to get
>     > > some identification from pipeline in each second or in  some
>     periodic
>     > > manner.)
>     >
>     > You could use gst_clock_new_periodic_id() together with a
>     > gst_clock_id_wait() or
>     > gst_clock_id_wait_async()
>     >
>     > Stefan
>     >
>     > > --
>     > > Sreerenj B
>     > > [hidden email] <mailto:[hidden email]>
>     <mailto:[hidden email] <mailto:[hidden email]>>
>     > > mob: +91 9739469496
>     > >
>     > >
>     >
>
>  
>
>   Hi ,
>
>    I captured the GstClockID of the pipeline in streaming-server
>
>  *id=gst_clock_new_periodic_id(media->media->pipeline->clock,start_time,interval);
>
> *And  then setted up the asynchronous  callback using the following.
>
> *gst_clock_id_wait_async(id,(GstClockCallback *)TickCallback,"Tick
> sending");*
>
> but  evenif i "stopped" or "paused" the client side player , i am
> getting the callback " TickCallback" in the server side on specified
> intervals!!!!!!! I take the clock of the pipeline currently running, but
> why i am getting this call back even i stopped the pipeline?Is there any
> way to rectify this?
>
> I setted up these settings in  
> *gst-rtsp-server/gst/rtsp-server/rtsp-client.c* ...inside the
> *handle_play()* function.
>
>
>     sreerenj
>
>
At first please at least edit the digest when replying. It would be nice if you
could turn off the digets feature for your subscription if you can participating
the discussions or ask questions.

If you set the pipeline to READY or NULL, the callback should not be called
anymore. Are you sure its still called? If so try to make a small example and if
its reproducable tehre , file a  bug and attach the example.

Stefan

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