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gstreamer-embedded — GStreamer on embedded platforms
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From: Felipe C. <fel...@gm...> - 2008-07-28 11:51:34
|
On Thu, Jul 17, 2008 at 1:59 PM, <jf...@so...> wrote: > Hi, all, > > I am doing a project to port the gstreamer framework to a target board, > which is running linux 2.6.24 on ARM 9, and the RAM on the board is 32Mbyte. > > When I tried to use the gstmad and alsalink plugin to play a MP3 file > ( "gst-launch location = abc.mp3 ! mad ! alsasink"), I found there is > constant audio underrrun reported from the ALSA driver and thus there are > on/off in the audio output. I've also tried other MP3 demux/decoders such as > ffdemux_mp3 and ffdec_mp3 but the result is same. After I enabled the > debugging on the filesrc plugin, I noticed there is a break/pause sometimes > (happening randomly) during the playback. I suspect this could be the reason > of the audio underrun but I don't know why this break/pause happens. > > Anyone has any idea? Please help. You might want to try adding mp3parse before the decoder, but my bet is on the ALSA driver. Best regards. -- Felipe Contreras |
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From: Zhao Liang-E. <E3...@mo...> - 2008-07-25 01:05:18
|
gstreamer doesn't have dependency on any GUI system, it just depends on glib, you can integrate it into any platforms with glib supported. I think all you need to do is to find the correct plugins or elements you want and create pipelines. about doc, you can refer http://gstreamer.freedesktop.org/documentation/ Zhao Liang ________________________________ From: gst...@li... [mailto:gst...@li...] On Behalf Of Dragos Cirjan Sent: Wednesday, July 16, 2008 5:34 PM To: gst...@li... Subject: [gst-embedded] using gstreamer in gnome and QT Hi there, 1st let me tell you that I'm very young with linux programming, yet I'm willing to learn. At this very moment GStreamer seems a huge nebula for me, that why I really need your help. I need to integrate GStreamer in a GNOME Window and in QT (QT4 would be great :D), actually I need to integrate it not in a simple window, but in smth similar to panels, because I need to make a huge interface to support up to 60 video streams in the same time. Can you please give me some simple examples, or some links to read about ? Thanks in advance, Chris -- ----------------------------------------------------------------- Cristian - Dragos, Cirjan ----------------------------------------------------------------- Email: dra...@ya... Email: dra...@it..., do...@bo... Telefon: +40726355762 |
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From: Dragos C. <dra...@gm...> - 2008-07-22 20:12:02
|
On Tue, Jul 22, 2008 at 5:07 PM, Benoit Fouet <ben...@pu...> wrote: > > Can anyone please help me deal with mjpeg. I noticed that the format > > it's supported, but I don't know what plugins to use. > > I guess jpegdec should be enough to handle mjpeg files<http://www.purplelabs.com> > Hey there. Thanks for the hint. I am planing to use gst for ip cameras. And some devices have mjpeg but others have only jpeg. (mjpeg is a stream, jpeg is static, so I have to call the http server again and again and again so i can update the info). Will jpegdec help me for jpeg also, or only for mjpeg? Thanks in advance. -- ----------------------------------------------------------------- Cristian - Dragos, Cirjan ----------------------------------------------------------------- Email: dra...@ya... Email: dra...@it..., do...@bo... Telefon: +40726355762 |
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From: Danilo F. <xh...@gm...> - 2008-07-22 14:33:10
|
Hi, look this: http://gstreamer.freedesktop.org/data/doc/gstreamer/head/pwg/html/section-iface-xoverlay.html Basically, you will create your window, get its XId, and tell your plugin (playbin will work) to use it. So, for each panel you will have a pipeline. []s On Wed, Jul 16, 2008 at 6:33 AM, Dragos Cirjan <dra...@gm...> wrote: > > Hi there, > > 1st let me tell you that I'm very young with linux programming, yet I'm > willing to learn. > At this very moment GStreamer seems a huge nebula for me, that why I really > need your help. > > I need to integrate GStreamer in a GNOME Window and in QT (QT4 would be > great :D), actually I need to integrate it not in a simple window, but in > smth similar to panels, because I need to make a huge interface to support > up to 60 video streams in the same time. > > Can you please give me some simple examples, or some links to read about ? > > Thanks in advance, > Chris > > -- > ----------------------------------------------------------------- > Cristian - Dragos, Cirjan > ----------------------------------------------------------------- > Email: dra...@ya... > Email: dra...@it..., do...@bo... > Telefon: +40726355762 > > ------------------------------------------------------------------------- > 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 > gst...@li... > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Danilo Freire Laboratório de Sistemas Embarcados e Computação Pervasiva Centro de Engenharia Elétrica e Informática - CEEI Universidade Federal de Campina Grande - UFCG |
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From: Benoit F. <ben...@pu...> - 2008-07-22 14:07:10
|
Hi, Dragos Cirjan wrote: > > Hi there. I'm running a Debian Lenny and a Suse 10.3 (to be updated to > Suse 11). > > Can anyone please help me deal with mjpeg. I noticed that the format > it's supported, but I don't know what plugins to use. I guess jpegdec should be enough to handle mjpeg files -- Benoit Fouet Purple Labs S.A. www.purplelabs.com |
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From: Dragos C. <dra...@gm...> - 2008-07-17 13:46:33
|
Hi there. I'm running a Debian Lenny and a Suse 10.3 (to be updated to Suse 11). Can anyone please help me deal with mjpeg. I noticed that the format it's supported, but I don't know what plugins to use. Thanks in advance. -- ----------------------------------------------------------------- Cristian - Dragos, Cirjan ----------------------------------------------------------------- Email: dra...@ya... Email: dra...@it..., do...@bo... Telefon: +40726355762 |
|
From: Dragos C. <dra...@gm...> - 2008-07-16 09:33:34
|
Hi there, 1st let me tell you that I'm very young with linux programming, yet I'm willing to learn. At this very moment GStreamer seems a huge nebula for me, that why I really need your help. I need to integrate GStreamer in a GNOME Window and in QT (QT4 would be great :D), actually I need to integrate it not in a simple window, but in smth similar to panels, because I need to make a huge interface to support up to 60 video streams in the same time. Can you please give me some simple examples, or some links to read about ? Thanks in advance, Chris -- ----------------------------------------------------------------- Cristian - Dragos, Cirjan ----------------------------------------------------------------- Email: dra...@ya... Email: dra...@it..., do...@bo... Telefon: +40726355762 |
|
From: Felipe C. <fel...@no...> - 2008-06-24 11:28:35
|
Hi everybody, The third pre-release of gst-openmax is ready: http://gstreamer.freedesktop.org/src/gst-openmax/pre/ gst-openmax is a GStreamer plug-in that allows communication with OpenMAX IL components. OpenMAX IL is an industry standard that provides an abstraction layer for computer graphics, video, and sound routines. It has been pushed specially by key industry players in embedded systems. For more information please visit: http://freedesktop.org/wiki/GstOpenMAX New component wrappers: * WMV dec * H.263, H.264 enc * AAC, AMR-NB, AMR-WB enc * ADPCM, G.711, G.729, iLBC enc/dec Among the important changes is that now the most important element (base-filter) has been redesigned to fix a lot of possible threading issues. Along with this change comes the addition of unit tests, for which a dummy OpenMAX IL implementation was created. Also many bugfixes all over the place. The plan is to do a couple more bug-fixes, possibly merge the tunneling branch (thanks to Frederik Vernelen from NXP) and the aim for the first release, for which if I understand correctly documentation and translation stuff is needed. Bugs resolved: * 527125: OMX AAC decoder should accept MPEG versions 2 and 4 * 517185: queue not emptied Contributors to this release: * Stefan Kost -- The GstOpenMAX team |
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From: Zhao Liang-E. <E3...@mo...> - 2008-06-24 01:35:04
|
Another walkaround is changing configure.ac, disable printf extension support.
Best Regards
Zhao Liang
-----Original Message-----
From: gst...@li... [mailto:gst...@li...] On Behalf Of Liu, Bin
Sent: Monday, June 23, 2008 10:42 PM
To: gst...@li...; gst...@li...
Subject: Re: [gst-embedded] [gst-devel] gstreamer segfault on ARM
I had the same issue. I manually modified gstconfig.h to make it work. Glib is out of my control.
-----Original Message-----
From: gst...@li... [mailto:gst...@li...] On Behalf Of Zhao Liang-E3423C
Sent: Monday, June 23, 2008 2:34 AM
To: Shi Ling-w20230; Bernard Blackham; gst...@li...; gst...@li...
Subject: Re: [gst-devel] [gst-embedded] gstreamer segfault on ARM
Yes, It is same issue with us, but we are using another solution, I can try his fix .
Best Regards
Zhao Liang 赵 亮
Tel:86-10-84733698
No.1 Wang Jing East Road, Chao Yang District, Beijing, China 100102
北京市朝阳区望京东路1号, 100102
-----Original Message-----
From: Shi Ling-w20230
Sent: Monday, June 23, 2008 3:29 PM
To: Bernard Blackham; gst...@li...; gst...@li...; Zhao Liang-E3423C
Subject: RE: [gst-devel] [gst-embedded] gstreamer segfault on ARM
Zhao Liang,
I remember we meet the same issue before. Could you check?
Shi Ling
Tel:86-10-84733539
Motorola (China) Technology Ltd.
No.1 Wang Jing East Road, Chao Yang District, 100102 Beijing
-----Original Message-----
From: gst...@li... [mailto:gst...@li...] On Behalf Of Bernard Blackham
Sent: Friday, June 20, 2008 3:30 PM
To: gst...@li...; gst...@li...
Subject: Re: [gst-devel] [gst-embedded] gstreamer segfault on ARM
[Taking to gst-devel]
Bernard Blackham wrote:
> Running a rather simple program:
>
> #include <gst/gst.h>
> int main() {
> gst_init(0, NULL);
> g_pipeline_new("pipeline");
> return 0;
> }
>
> with GST_DEBUG=4 causes the program to segfault when it goes to print
> out one of its trace messages.
I tracked down the cause of this to an incompatibility with my build of glib. When cross-compiling glib, it decided to not like glibc's printf (because it tried running some printf tests and failed, because it was cross-compiling), and thus went and used its own printf implementation.
Gstreamer was not aware of this decision in the glib build though, so it happily went ahead and used glib assuming it had a glibc printf that could support printf extensions (that it uses to format %P in strings).
Rebuilding glib with the right configure cache settings (below) solved the gstreamer crash.
glib_cv_long_long_format=ll
ac_cv_func_printf_unix98=yes
ac_cv_func_vsnprintf_c99=yes
So this solved the problem, but I'm wondering if there should be a better way for gstreamer to detect if it is safe to use %P or not: not only does glibc need to provide it, but glib needs to be using it too.
Or is the fact that glibc provides it enough proof that glib should be using it and anything else is a broken setup like mine was?
TIA,
Bernard.
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From: Liu, B. <b-...@ti...> - 2008-06-23 14:41:58
|
I had the same issue. I manually modified gstconfig.h to make it work. Glib is out of my control.
-----Original Message-----
From: gst...@li... [mailto:gst...@li...] On Behalf Of Zhao Liang-E3423C
Sent: Monday, June 23, 2008 2:34 AM
To: Shi Ling-w20230; Bernard Blackham; gst...@li...; gst...@li...
Subject: Re: [gst-devel] [gst-embedded] gstreamer segfault on ARM
Yes, It is same issue with us, but we are using another solution, I can try his fix .
Best Regards
Zhao Liang 赵 亮
Tel:86-10-84733698
No.1 Wang Jing East Road, Chao Yang District, Beijing, China 100102
北京市朝阳区望京东路1号, 100102
-----Original Message-----
From: Shi Ling-w20230
Sent: Monday, June 23, 2008 3:29 PM
To: Bernard Blackham; gst...@li...; gst...@li...; Zhao Liang-E3423C
Subject: RE: [gst-devel] [gst-embedded] gstreamer segfault on ARM
Zhao Liang,
I remember we meet the same issue before. Could you check?
Shi Ling
Tel:86-10-84733539
Motorola (China) Technology Ltd.
No.1 Wang Jing East Road, Chao Yang District, 100102 Beijing
-----Original Message-----
From: gst...@li... [mailto:gst...@li...] On Behalf Of Bernard Blackham
Sent: Friday, June 20, 2008 3:30 PM
To: gst...@li...; gst...@li...
Subject: Re: [gst-devel] [gst-embedded] gstreamer segfault on ARM
[Taking to gst-devel]
Bernard Blackham wrote:
> Running a rather simple program:
>
> #include <gst/gst.h>
> int main() {
> gst_init(0, NULL);
> g_pipeline_new("pipeline");
> return 0;
> }
>
> with GST_DEBUG=4 causes the program to segfault when it goes to print
> out one of its trace messages.
I tracked down the cause of this to an incompatibility with my build of glib. When cross-compiling glib, it decided to not like glibc's printf (because it tried running some printf tests and failed, because it was cross-compiling), and thus went and used its own printf implementation.
Gstreamer was not aware of this decision in the glib build though, so it happily went ahead and used glib assuming it had a glibc printf that could support printf extensions (that it uses to format %P in strings).
Rebuilding glib with the right configure cache settings (below) solved the gstreamer crash.
glib_cv_long_long_format=ll
ac_cv_func_printf_unix98=yes
ac_cv_func_vsnprintf_c99=yes
So this solved the problem, but I'm wondering if there should be a better way for gstreamer to detect if it is safe to use %P or not: not only does glibc need to provide it, but glib needs to be using it too.
Or is the fact that glibc provides it enough proof that glib should be using it and anything else is a broken setup like mine was?
TIA,
Bernard.
-------------------------------------------------------------------------
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It's the best place to buy or sell services for just about anything Open Source.
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_______________________________________________
gstreamer-devel mailing list
gst...@li...
https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
|
|
From: Zhao Liang-E. <E3...@mo...> - 2008-06-23 07:33:53
|
Yes, It is same issue with us, but we are using another solution, I can try his fix .
Best Regards
Zhao Liang 赵 亮
Tel:86-10-84733698
No.1 Wang Jing East Road, Chao Yang District, Beijing, China 100102
北京市朝阳区望京东路1号, 100102
-----Original Message-----
From: Shi Ling-w20230
Sent: Monday, June 23, 2008 3:29 PM
To: Bernard Blackham; gst...@li...; gst...@li...; Zhao Liang-E3423C
Subject: RE: [gst-devel] [gst-embedded] gstreamer segfault on ARM
Zhao Liang,
I remember we meet the same issue before. Could you check?
Shi Ling
Tel:86-10-84733539
Motorola (China) Technology Ltd.
No.1 Wang Jing East Road, Chao Yang District, 100102 Beijing
-----Original Message-----
From: gst...@li... [mailto:gst...@li...] On Behalf Of Bernard Blackham
Sent: Friday, June 20, 2008 3:30 PM
To: gst...@li...; gst...@li...
Subject: Re: [gst-devel] [gst-embedded] gstreamer segfault on ARM
[Taking to gst-devel]
Bernard Blackham wrote:
> Running a rather simple program:
>
> #include <gst/gst.h>
> int main() {
> gst_init(0, NULL);
> g_pipeline_new("pipeline");
> return 0;
> }
>
> with GST_DEBUG=4 causes the program to segfault when it goes to print
> out one of its trace messages.
I tracked down the cause of this to an incompatibility with my build of glib. When cross-compiling glib, it decided to not like glibc's printf (because it tried running some printf tests and failed, because it was cross-compiling), and thus went and used its own printf implementation.
Gstreamer was not aware of this decision in the glib build though, so it happily went ahead and used glib assuming it had a glibc printf that could support printf extensions (that it uses to format %P in strings).
Rebuilding glib with the right configure cache settings (below) solved the gstreamer crash.
glib_cv_long_long_format=ll
ac_cv_func_printf_unix98=yes
ac_cv_func_vsnprintf_c99=yes
So this solved the problem, but I'm wondering if there should be a better way for gstreamer to detect if it is safe to use %P or not: not only does glibc need to provide it, but glib needs to be using it too.
Or is the fact that glibc provides it enough proof that glib should be using it and anything else is a broken setup like mine was?
TIA,
Bernard.
-------------------------------------------------------------------------
Check out the new SourceForge.net Marketplace.
It's the best place to buy or sell services for just about anything Open Source.
http://sourceforge.net/services/buy/index.php
_______________________________________________
gstreamer-devel mailing list
gst...@li...
https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
|
|
From: Shi Ling-w. <li...@mo...> - 2008-06-23 07:28:59
|
Zhao Liang,
I remember we meet the same issue before. Could you check?
Shi Ling
Tel:86-10-84733539
Motorola (China) Technology Ltd.
No.1 Wang Jing East Road, Chao Yang District, 100102 Beijing
-----Original Message-----
From: gst...@li...
[mailto:gst...@li...] On Behalf Of
Bernard Blackham
Sent: Friday, June 20, 2008 3:30 PM
To: gst...@li...;
gst...@li...
Subject: Re: [gst-devel] [gst-embedded] gstreamer segfault on ARM
[Taking to gst-devel]
Bernard Blackham wrote:
> Running a rather simple program:
>
> #include <gst/gst.h>
> int main() {
> gst_init(0, NULL);
> g_pipeline_new("pipeline");
> return 0;
> }
>
> with GST_DEBUG=4 causes the program to segfault when it goes to
> print out one of its trace messages.
I tracked down the cause of this to an incompatibility with my build of
glib. When cross-compiling glib, it decided to not like glibc's printf
(because it tried running some printf tests and failed, because it was
cross-compiling), and thus went and used its own printf implementation.
Gstreamer was not aware of this decision in the glib build though, so it
happily went ahead and used glib assuming it had a glibc printf that
could support printf extensions (that it uses to format %P in strings).
Rebuilding glib with the right configure cache settings (below) solved
the gstreamer crash.
glib_cv_long_long_format=ll
ac_cv_func_printf_unix98=yes
ac_cv_func_vsnprintf_c99=yes
So this solved the problem, but I'm wondering if there should be a
better way for gstreamer to detect if it is safe to use %P or not: not
only does glibc need to provide it, but glib needs to be using it too.
Or is the fact that glibc provides it enough proof that glib should be
using it and anything else is a broken setup like mine was?
TIA,
Bernard.
------------------------------------------------------------------------
-
Check out the new SourceForge.net Marketplace.
It's the best place to buy or sell services for
just about anything Open Source.
http://sourceforge.net/services/buy/index.php
_______________________________________________
gstreamer-devel mailing list
gst...@li...
https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
|
|
From: Stefan K. <en...@ho...> - 2008-06-22 05:56:39
|
hi, you can use multiqueue, which is severl synchronized queues. But I belive you problem needs to be fixed elsewhere. Can you tell us how the whole pipeline looks like? You said your aac-decoder does not set timestamps? Is it putting GST_CLOCK_TIME_NONE there (don't use 0)? You should implement qos and timestamps. The sink will tell you whats the next expected timestamp and your aac decoder could skip packets that come late already. Stefan Shenhong Wang schrieb: > Thanks! Brad. > However I use two queues for audio and video separately but one > pipeline. So it would be impossible for me to pause the pipeline? > because the application can play video very well even the audio is blocked. > Why the alsasink will drop all packets(frames) after a break or so? > thanks again > > Shenhong > > > > > ------------------------------------------------------------------------ > Subject: RE: [gst-embedded] Question on gst_plugin alsasink > Date: Wed, 18 Jun 2008 16:55:38 +0800 > From: bi...@mo... > To: E3...@mo...; qc...@ho...; > gst...@li... > > > > yes, you can refernce how to use queue. you can set water mark in > queue.And then post message to bus if lower than mater mark. in your > main app you can recieve the message to pause the pipeline. > > if higher water mark, you can use the same mechanism. > > > > > ------------------------------------------------------------------------ > *From:* gst...@li... > [mailto:gst...@li...] *On Behalf > Of *Zhao Liang-E3423C > *Sent:* Wednesday, June 18, 2008 4:49 PM > *To:* Shenhong Wang; gst...@li... > *Subject:* Re: [gst-embedded] Question on gst_plugin alsasink > > Hi shenhong, > > A simply solution you can try. > > Put a queue before alsasink, when queue is dry, pause pipeline, and > restart pipeline when queue bufferred enough data. > > > > *Best Regards > Zhao Liang * > > ------------------------------------------------------------------------ > *From:* Shenhong Wang [mailto:qc...@ho...] > *Sent:* Wednesday, June 18, 2008 4:44 PM > *To:* Zhao Liang-E3423C; gst...@li... > *Subject:* RE: [gst-embedded] Question on gst_plugin alsasink > > Hi, Zhao Liang: > Generally, the aacdec &alsasink will not play out any audio > frames(packets) after its source element has a break to send audio > frames (packets) to them. It looks the alsasink drops all > frames(packets) from the break. The break is needed because we have > more video frames and sometime the wireless signal is not good. > It looks the aacdec is slower than the expectation from alsasink.If > so, how to fix the issue? thanks! > > best Regards! > Shenhong > > > > > > > > > ------------------------------------------------------------------------ > Subject: RE: [gst-embedded] Question on gst_plugin alsasink > Date: Wed, 18 Jun 2008 14:29:27 +0800 > From: E3...@mo... > To: qc...@ho...; gst...@li... > > Hi Shenhong, > > Your issue is very similar with the issue I even met. I think it > is due to gstbaseaudiosink/gstaudiosink, it will drop the > packets by gstringbuffer when read rate is bigger than write > rate in ringbuffer, please see gstringbuffer.c > gst_ring_buffer_commit_full (). > > For the rootcause, I think maybe the alsasink audiodevice buffer > is too big or your aac decoder is too slow. > > > *Best Regards > Zhao Liang* > > ------------------------------------------------------------------------ > *From:* gst...@li... > [mailto:gst...@li...] *On > Behalf Of *Shenhong Wang > *Sent:* Wednesday, June 18, 2008 2:21 PM > *To:* gst...@li... > *Subject:* [gst-embedded] Question on gst_plugin alsasink > > > Dear all, > Now we are using alsasink to play audio on Marvell PXA310 board. > The audio is aac format. The audio frames(packets) > are frequently sent to the aac decoder & alsasink to play out. > Unfortunately only the begining frames can be played out and > then nothing is played out. > If we save those audio frames into a file, the aac > decoder&alsasink can be successfully played out. It means the > audio frames are ok. > Could anyone tell me what's the difference for alsasink to > process audio packets and files? How to fix the above issue? > thank you very much! > > Best Regards! > Shenhong WANG > > ------------------------------------------------------------------------ > Connect to the next generation of MSN Messenger Get it now! > <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline> > > > ------------------------------------------------------------------------ > Connect to the next generation of MSN Messenger Get it now! > <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline> > > > ------------------------------------------------------------------------ > Connect to the next generation of MSN Messenger Get it now! > <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline> > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > Check out the new SourceForge.net Marketplace. > It's the best place to buy or sell services for > just about anything Open Source. > http://sourceforge.net/services/buy/index.php > > > ------------------------------------------------------------------------ > > _______________________________________________ > Gstreamer-embedded mailing list > Gst...@li... > https://lists.sourceforge.net/lists/listinfo/gstreamer-embedded |
|
From: Tim M. <tim...@co...> - 2008-06-20 08:48:50
|
On Fri, 2008-06-20 at 02:48 +0000, Shenhong Wang wrote: > I read chapter 14th of the Plugin Writer's guide again > http://gstreamer.freedesktop.org/data/doc/gstreamer/head/pwg/html/section-time-data-flow.html > but confused with the sentence: > "First, the source element sends a discontinous event. This event > carries information about the current relative time of the next > sample. ". Looks like you found a chapter that hasn't been updated since the GStreamer-0.8 days. You should ignore everything in there, it doesn't apply any longer. (Please file a bug against the documentation for this, if there isn't one already) > Can anyone of you tell me how to send the discontinuous event in our > source/parser element? thanks a lot! In GStreamer-0.10 you need to send a NEWSEGMENT event instead. See the design docs (in the gstreamer core source code under docs/design/) and the API documentation for gst_event_new_new_segment_full() [1] for more information. Cheers -Tim [1] http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstEvent.html#gst-event-new-new-segment-full |
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From: Bernard B. <be...@la...> - 2008-06-20 07:29:27
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[Taking to gst-devel]
Bernard Blackham wrote:
> Running a rather simple program:
>
> #include <gst/gst.h>
> int main() {
> gst_init(0, NULL);
> g_pipeline_new("pipeline");
> return 0;
> }
>
> with GST_DEBUG=4 causes the program to segfault when it goes to
> print out one of its trace messages.
I tracked down the cause of this to an incompatibility with my build of
glib. When cross-compiling glib, it decided to not like glibc's printf
(because it tried running some printf tests and failed, because it was
cross-compiling), and thus went and used its own printf implementation.
Gstreamer was not aware of this decision in the glib build though, so it
happily went ahead and used glib assuming it had a glibc printf that
could support printf extensions (that it uses to format %P in strings).
Rebuilding glib with the right configure cache settings (below) solved
the gstreamer crash.
glib_cv_long_long_format=ll
ac_cv_func_printf_unix98=yes
ac_cv_func_vsnprintf_c99=yes
So this solved the problem, but I'm wondering if there should be a
better way for gstreamer to detect if it is safe to use %P or not: not
only does glibc need to provide it, but glib needs to be using it too.
Or is the fact that glibc provides it enough proof that glib should be
using it and anything else is a broken setup like mine was?
TIA,
Bernard.
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From: Shenhong W. <qc...@ho...> - 2008-06-20 02:48:38
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Dear all, With your great help, now we may play out audio&video and we need add clock/timestamp for our media player. I read chapter 14th of the Plugin Writer's guider again http://gstreamer.freedesktop.org/data/doc/gstreamer/head/pwg/html/section-time-data-flow.html but confused with the sentence: "First, the source element sends a discontinous event. This event carries information about the current relative time of the next sample. ". Can anyone of you tell me how to send the discontinuous event in our source/parser element? thanks a lot! Best Regards! Shenhong _________________________________________________________________ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE |
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From: Bernard B. <be...@la...> - 2008-06-19 12:24:49
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Running a rather simple program:
#include <gst/gst.h>
int main() {
gst_init(0, NULL);
g_pipeline_new("pipeline");
return 0;
}
with GST_DEBUG=4 causes the program to segfault when it goes to
print out one of its trace messages.
Can someone else please try and reproduce this? Simply, compile the
above program, and run:
$ GST_DEBUG=4 ./test
I've tracked it down to this line of code in gstbin.c's
gst_bin_init:
479 GST_DEBUG_OBJECT (bin, "using bus %" GST_PTR_FORMAT " to listen to children",
480 bus);
Digging deeper, what happens is that in gst_debug_log_default, the
arguments turn to rubbish, somewhere along the way.
I'm using gstreamer 0.10.20, glib 2.16.3 and gcc 4.1.1 - I suspect a
compiler bug, but I'd like to see if anybody else can reproduce this
before I go rebuilding toolchains :)
Cheers,
Bernard.
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From: Zhao Liang-E. <E3...@mo...> - 2008-06-19 05:08:21
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please check queue signals "underrun" "overrun" .... Zhao Liang ________________________________ From: Shenhong Wang [mailto:qc...@ho...] Sent: Thursday, June 19, 2008 9:47 AM To: Zhao Bin-E6223C; Zhao Liang-E3423C; gst...@li... Subject: RE: [gst-embedded] Question on gst_plugin alsasink Hi, Brad or Zhao Liang: Is it possible for you to publish an example - how to post a message to bus and pause/play pipeline? thanks a lot! Best Regards! Shenhong ________________________________ Subject: RE: [gst-embedded] Question on gst_plugin alsasink Date: Wed, 18 Jun 2008 17:08:09 +0800 From: bi...@mo... To: qc...@ho...; E3...@mo...; gst...@li... I think it is due to gstbaseaudiosink/gstaudiosink, it will drop the packets by gstringbuffer when read rate is bigger than write rate in ringbuffer, please see gstringbuffer.c gst_ring_buffer_commit_full (). Please check code in gstbaseaudiosink.c and gstaudiosink.c i remember the sig_write is lower than sig_done,sink will drop the buffer. ________________________________ From: Shenhong Wang [mailto:qc...@ho...] Sent: Wednesday, June 18, 2008 5:05 PM To: Zhao Bin-E6223C; Zhao Liang-E3423C; gst...@li... Subject: RE: [gst-embedded] Question on gst_plugin alsasink Thanks! Brad. However I use two queues for audio and video separately but one pipeline. So it would be impossible for me to pause the pipeline? because the application can play video very well even the audio is blocked. Why the alsasink will drop all packets(frames) after a break or so? thanks again Shenhong ________________________________ Subject: RE: [gst-embedded] Question on gst_plugin alsasink Date: Wed, 18 Jun 2008 16:55:38 +0800 From: bi...@mo... To: E3...@mo...; qc...@ho...; gst...@li... yes, you can refernce how to use queue. you can set water mark in queue.And then post message to bus if lower than mater mark. in your main app you can recieve the message to pause the pipeline. if higher water mark, you can use the same mechanism. ________________________________ From: gst...@li... [mailto:gst...@li...] On Behalf Of Zhao Liang-E3423C Sent: Wednesday, June 18, 2008 4:49 PM To: Shenhong Wang; gst...@li... Subject: Re: [gst-embedded] Question on gst_plugin alsasink Hi shenhong, A simply solution you can try. Put a queue before alsasink, when queue is dry, pause pipeline, and restart pipeline when queue bufferred enough data. Best Regards Zhao Liang ________________________________ From: Shenhong Wang [mailto:qc...@ho...] Sent: Wednesday, June 18, 2008 4:44 PM To: Zhao Liang-E3423C; gst...@li... Subject: RE: [gst-embedded] Question on gst_plugin alsasink Hi, Zhao Liang: Generally, the aacdec &alsasink will not play out any audio frames(packets) after its source element has a break to send audio frames (packets) to them. It looks the alsasink drops all frames(packets) from the break. The break is needed because we have more video frames and sometime the wireless signal is not good. It looks the aacdec is slower than the expectation from alsasink.If so, how to fix the issue? thanks! best Regards! Shenhong ________________________________ Subject: RE: [gst-embedded] Question on gst_plugin alsasink Date: Wed, 18 Jun 2008 14:29:27 +0800 From: E3...@mo... To: qc...@ho...; gst...@li... Hi Shenhong, Your issue is very similar with the issue I even met. I think it is due to gstbaseaudiosink/gstaudiosink, it will drop the packets by gstringbuffer when read rate is bigger than write rate in ringbuffer, please see gstringbuffer.c gst_ring_buffer_commit_full (). For the rootcause, I think maybe the alsasink audiodevice buffer is too big or your aac decoder is too slow. Best Regards Zhao Liang ________________________________ From: gst...@li... [mailto:gst...@li...] On Behalf Of Shenhong Wang Sent: Wednesday, June 18, 2008 2:21 PM To: gst...@li... Subject: [gst-embedded] Question on gst_plugin alsasink Dear all, Now we are using alsasink to play audio on Marvell PXA310 board. The audio is aac format. The audio frames(packets) are frequently sent to the aac decoder & alsasink to play out. Unfortunately only the begining frames can be played out and then nothing is played out. If we save those audio frames into a file, the aac decoder&alsasink can be successfully played out. It means the audio frames are ok. Could anyone tell me what's the difference for alsasink to process audio packets and files? How to fix the above issue? thank you very much! Best Regards! Shenhong WANG ________________________________ Connect to the next generation of MSN Messenger Get it now! <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&sou rce=wlmailtagline> ________________________________ Connect to the next generation of MSN Messenger Get it now! <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&sou rce=wlmailtagline> ________________________________ Connect to the next generation of MSN Messenger Get it now! <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&sou rce=wlmailtagline> ________________________________ Get news, entertainment and everything you care about at Live.com. Check it out! <http://www.live.com/getstarted.aspx> |
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From: Shenhong W. <qc...@ho...> - 2008-06-19 01:46:41
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Hi, Brad or Zhao Liang: Is it possible for you to publish an example - how to post a message to bus and pause/play pipeline? thanks a lot! Best Regards! Shenhong Subject: RE: [gst-embedded] Question on gst_plugin alsasinkDate: Wed, 18 Jun 2008 17:08:09 +0800From: bi...@mo...To: qc...@ho...; E3...@mo...; gst...@li... I think it is due to gstbaseaudiosink/gstaudiosink, it will drop the packets by gstringbuffer when read rate is bigger than write rate in ringbuffer, please see gstringbuffer.c gst_ring_buffer_commit_full (). Please check code in gstbaseaudiosink.c and gstaudiosink.c i remember the sig_write is lower than sig_done,sink will drop the buffer. From: Shenhong Wang [mailto:qc...@ho...] Sent: Wednesday, June 18, 2008 5:05 PMTo: Zhao Bin-E6223C; Zhao Liang-E3423C; gst...@li...Subject: RE: [gst-embedded] Question on gst_plugin alsasink Thanks! Brad.However I use two queues for audio and video separately but one pipeline. So it would be impossible for me to pause the pipeline? because the application can play video very well even the audio is blocked. Why the alsasink will drop all packets(frames) after a break or so? thanks again Shenhong Subject: RE: [gst-embedded] Question on gst_plugin alsasinkDate: Wed, 18 Jun 2008 16:55:38 +0800From: bi...@mo...To: E3...@mo...; qc...@ho...; gst...@li... yes, you can refernce how to use queue. you can set water mark in queue.And then post message to bus if lower than mater mark. in your main app you can recieve the message to pause the pipeline. if higher water mark, you can use the same mechanism. From: gst...@li... [mailto:gst...@li...] On Behalf Of Zhao Liang-E3423CSent: Wednesday, June 18, 2008 4:49 PMTo: Shenhong Wang; gst...@li...Subject: Re: [gst-embedded] Question on gst_plugin alsasink Hi shenhong, A simply solution you can try. Put a queue before alsasink, when queue is dry, pause pipeline, and restart pipeline when queue bufferred enough data. Best RegardsZhao Liang From: Shenhong Wang [mailto:qc...@ho...] Sent: Wednesday, June 18, 2008 4:44 PMTo: Zhao Liang-E3423C; gst...@li...Subject: RE: [gst-embedded] Question on gst_plugin alsasink Hi, Zhao Liang:Generally, the aacdec &alsasink will not play out any audio frames(packets) after its source element has a break to send audio frames (packets) to them. It looks the alsasink drops all frames(packets) from the break. The break is needed because we have more video frames and sometime the wireless signal is not good. It looks the aacdec is slower than the expectation from alsasink.If so, how to fix the issue? thanks! best Regards!Shenhong Subject: RE: [gst-embedded] Question on gst_plugin alsasinkDate: Wed, 18 Jun 2008 14:29:27 +0800From: E3...@mo...To: qc...@ho...; gst...@li... Hi Shenhong, Your issue is very similar with the issue I even met. I think it is due to gstbaseaudiosink/gstaudiosink, it will drop the packets by gstringbuffer when read rate is bigger than write rate in ringbuffer, please see gstringbuffer.c gst_ring_buffer_commit_full (). For the rootcause, I think maybe the alsasink audiodevice buffer is too big or your aac decoder is too slow. Best RegardsZhao Liang From: gst...@li... [mailto:gst...@li...] On Behalf Of Shenhong WangSent: Wednesday, June 18, 2008 2:21 PMTo: gst...@li...Subject: [gst-embedded] Question on gst_plugin alsasink Dear all,Now we are using alsasink to play audio on Marvell PXA310 board. The audio is aac format. The audio frames(packets) are frequently sent to the aac decoder & alsasink to play out. Unfortunately only the begining frames can be played out and then nothing is played out. If we save those audio frames into a file, the aac decoder&alsasink can be successfully played out. It means the audio frames are ok. Could anyone tell me what's the difference for alsasink to process audio packets and files? How to fix the above issue? thank you very much! Best Regards!Shenhong WANG Connect to the next generation of MSN Messenger Get it now! Connect to the next generation of MSN Messenger Get it now! Connect to the next generation of MSN Messenger Get it now! _________________________________________________________________ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx |
|
From: Zhao Bin-E. <bi...@mo...> - 2008-06-18 09:08:13
|
I think it is due to gstbaseaudiosink/gstaudiosink, it will drop the packets by gstringbuffer when read rate is bigger than write rate in ringbuffer, please see gstringbuffer.c gst_ring_buffer_commit_full (). Please check code in gstbaseaudiosink.c and gstaudiosink.c i remember the sig_write is lower than sig_done,sink will drop the buffer. ________________________________ From: Shenhong Wang [mailto:qc...@ho...] Sent: Wednesday, June 18, 2008 5:05 PM To: Zhao Bin-E6223C; Zhao Liang-E3423C; gst...@li... Subject: RE: [gst-embedded] Question on gst_plugin alsasink Thanks! Brad. However I use two queues for audio and video separately but one pipeline. So it would be impossible for me to pause the pipeline? because the application can play video very well even the audio is blocked. Why the alsasink will drop all packets(frames) after a break or so? thanks again Shenhong ________________________________ Subject: RE: [gst-embedded] Question on gst_plugin alsasink Date: Wed, 18 Jun 2008 16:55:38 +0800 From: bi...@mo... To: E3...@mo...; qc...@ho...; gst...@li... yes, you can refernce how to use queue. you can set water mark in queue.And then post message to bus if lower than mater mark. in your main app you can recieve the message to pause the pipeline. if higher water mark, you can use the same mechanism. ________________________________ From: gst...@li... [mailto:gst...@li...] On Behalf Of Zhao Liang-E3423C Sent: Wednesday, June 18, 2008 4:49 PM To: Shenhong Wang; gst...@li... Subject: Re: [gst-embedded] Question on gst_plugin alsasink Hi shenhong, A simply solution you can try. Put a queue before alsasink, when queue is dry, pause pipeline, and restart pipeline when queue bufferred enough data. Best Regards Zhao Liang ________________________________ From: Shenhong Wang [mailto:qc...@ho...] Sent: Wednesday, June 18, 2008 4:44 PM To: Zhao Liang-E3423C; gst...@li... Subject: RE: [gst-embedded] Question on gst_plugin alsasink Hi, Zhao Liang: Generally, the aacdec &alsasink will not play out any audio frames(packets) after its source element has a break to send audio frames (packets) to them. It looks the alsasink drops all frames(packets) from the break. The break is needed because we have more video frames and sometime the wireless signal is not good. It looks the aacdec is slower than the expectation from alsasink.If so, how to fix the issue? thanks! best Regards! Shenhong ________________________________ Subject: RE: [gst-embedded] Question on gst_plugin alsasink Date: Wed, 18 Jun 2008 14:29:27 +0800 From: E3...@mo... To: qc...@ho...; gst...@li... Hi Shenhong, Your issue is very similar with the issue I even met. I think it is due to gstbaseaudiosink/gstaudiosink, it will drop the packets by gstringbuffer when read rate is bigger than write rate in ringbuffer, please see gstringbuffer.c gst_ring_buffer_commit_full (). For the rootcause, I think maybe the alsasink audiodevice buffer is too big or your aac decoder is too slow. Best Regards Zhao Liang ________________________________ From: gst...@li... [mailto:gst...@li...] On Behalf Of Shenhong Wang Sent: Wednesday, June 18, 2008 2:21 PM To: gst...@li... Subject: [gst-embedded] Question on gst_plugin alsasink Dear all, Now we are using alsasink to play audio on Marvell PXA310 board. The audio is aac format. The audio frames(packets) are frequently sent to the aac decoder & alsasink to play out. Unfortunately only the begining frames can be played out and then nothing is played out. If we save those audio frames into a file, the aac decoder&alsasink can be successfully played out. It means the audio frames are ok. Could anyone tell me what's the difference for alsasink to process audio packets and files? How to fix the above issue? thank you very much! Best Regards! Shenhong WANG ________________________________ Connect to the next generation of MSN Messenger Get it now! <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&sou rce=wlmailtagline> ________________________________ Connect to the next generation of MSN Messenger Get it now! <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&sou rce=wlmailtagline> ________________________________ Connect to the next generation of MSN Messenger Get it now! <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&sou rce=wlmailtagline> |
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From: Shenhong W. <qc...@ho...> - 2008-06-18 09:04:45
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Thanks! Brad. However I use two queues for audio and video separately but one pipeline. So it would be impossible for me to pause the pipeline? because the application can play video very well even the audio is blocked. Why the alsasink will drop all packets(frames) after a break or so? thanks again Shenhong Subject: RE: [gst-embedded] Question on gst_plugin alsasinkDate: Wed, 18 Jun 2008 16:55:38 +0800From: bi...@mo...To: E3...@mo...; qc...@ho...; gst...@li... yes, you can refernce how to use queue. you can set water mark in queue.And then post message to bus if lower than mater mark. in your main app you can recieve the message to pause the pipeline. if higher water mark, you can use the same mechanism. From: gst...@li... [mailto:gst...@li...] On Behalf Of Zhao Liang-E3423CSent: Wednesday, June 18, 2008 4:49 PMTo: Shenhong Wang; gst...@li...Subject: Re: [gst-embedded] Question on gst_plugin alsasink Hi shenhong, A simply solution you can try. Put a queue before alsasink, when queue is dry, pause pipeline, and restart pipeline when queue bufferred enough data. Best RegardsZhao Liang From: Shenhong Wang [mailto:qc...@ho...] Sent: Wednesday, June 18, 2008 4:44 PMTo: Zhao Liang-E3423C; gst...@li...Subject: RE: [gst-embedded] Question on gst_plugin alsasink Hi, Zhao Liang:Generally, the aacdec &alsasink will not play out any audio frames(packets) after its source element has a break to send audio frames (packets) to them. It looks the alsasink drops all frames(packets) from the break. The break is needed because we have more video frames and sometime the wireless signal is not good. It looks the aacdec is slower than the expectation from alsasink.If so, how to fix the issue? thanks! best Regards!Shenhong Subject: RE: [gst-embedded] Question on gst_plugin alsasinkDate: Wed, 18 Jun 2008 14:29:27 +0800From: E3...@mo...To: qc...@ho...; gst...@li... Hi Shenhong, Your issue is very similar with the issue I even met. I think it is due to gstbaseaudiosink/gstaudiosink, it will drop the packets by gstringbuffer when read rate is bigger than write rate in ringbuffer, please see gstringbuffer.c gst_ring_buffer_commit_full (). For the rootcause, I think maybe the alsasink audiodevice buffer is too big or your aac decoder is too slow. Best RegardsZhao Liang From: gst...@li... [mailto:gst...@li...] On Behalf Of Shenhong WangSent: Wednesday, June 18, 2008 2:21 PMTo: gst...@li...Subject: [gst-embedded] Question on gst_plugin alsasink Dear all,Now we are using alsasink to play audio on Marvell PXA310 board. The audio is aac format. The audio frames(packets) are frequently sent to the aac decoder & alsasink to play out. Unfortunately only the begining frames can be played out and then nothing is played out. If we save those audio frames into a file, the aac decoder&alsasink can be successfully played out. It means the audio frames are ok. Could anyone tell me what's the difference for alsasink to process audio packets and files? How to fix the above issue? thank you very much! Best Regards!Shenhong WANG Connect to the next generation of MSN Messenger Get it now! Connect to the next generation of MSN Messenger Get it now! _________________________________________________________________ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline |
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From: Zhao Bin-E. <bi...@mo...> - 2008-06-18 08:55:39
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yes, you can refernce how to use queue. you can set water mark in queue.And then post message to bus if lower than mater mark. in your main app you can recieve the message to pause the pipeline. if higher water mark, you can use the same mechanism. ________________________________ From: gst...@li... [mailto:gst...@li...] On Behalf Of Zhao Liang-E3423C Sent: Wednesday, June 18, 2008 4:49 PM To: Shenhong Wang; gst...@li... Subject: Re: [gst-embedded] Question on gst_plugin alsasink Hi shenhong, A simply solution you can try. Put a queue before alsasink, when queue is dry, pause pipeline, and restart pipeline when queue bufferred enough data. Best Regards Zhao Liang ________________________________ From: Shenhong Wang [mailto:qc...@ho...] Sent: Wednesday, June 18, 2008 4:44 PM To: Zhao Liang-E3423C; gst...@li... Subject: RE: [gst-embedded] Question on gst_plugin alsasink Hi, Zhao Liang: Generally, the aacdec &alsasink will not play out any audio frames(packets) after its source element has a break to send audio frames (packets) to them. It looks the alsasink drops all frames(packets) from the break. The break is needed because we have more video frames and sometime the wireless signal is not good. It looks the aacdec is slower than the expectation from alsasink.If so, how to fix the issue? thanks! best Regards! Shenhong ________________________________ Subject: RE: [gst-embedded] Question on gst_plugin alsasink Date: Wed, 18 Jun 2008 14:29:27 +0800 From: E3...@mo... To: qc...@ho...; gst...@li... Hi Shenhong, Your issue is very similar with the issue I even met. I think it is due to gstbaseaudiosink/gstaudiosink, it will drop the packets by gstringbuffer when read rate is bigger than write rate in ringbuffer, please see gstringbuffer.c gst_ring_buffer_commit_full (). For the rootcause, I think maybe the alsasink audiodevice buffer is too big or your aac decoder is too slow. Best Regards Zhao Liang ________________________________ From: gst...@li... [mailto:gst...@li...] On Behalf Of Shenhong Wang Sent: Wednesday, June 18, 2008 2:21 PM To: gst...@li... Subject: [gst-embedded] Question on gst_plugin alsasink Dear all, Now we are using alsasink to play audio on Marvell PXA310 board. The audio is aac format. The audio frames(packets) are frequently sent to the aac decoder & alsasink to play out. Unfortunately only the begining frames can be played out and then nothing is played out. If we save those audio frames into a file, the aac decoder&alsasink can be successfully played out. It means the audio frames are ok. Could anyone tell me what's the difference for alsasink to process audio packets and files? How to fix the above issue? thank you very much! Best Regards! Shenhong WANG ________________________________ Connect to the next generation of MSN Messenger Get it now! <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&sou rce=wlmailtagline> ________________________________ Connect to the next generation of MSN Messenger Get it now! <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&sou rce=wlmailtagline> |
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From: Shenhong W. <qc...@ho...> - 2008-06-18 08:54:39
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Zhao Liang:Thanks! Now we use a queue before the aac decoder &alsasink. How to check the queue is empty and pause/restart pipeline? hehe...thanks! Best Regards! Shenhong Subject: RE: [gst-embedded] Question on gst_plugin alsasinkDate: Wed, 18 Jun 2008 16:49:08 +0800From: E3...@mo...To: qc...@ho...; gst...@li... Hi shenhong, A simply solution you can try. Put a queue before alsasink, when queue is dry, pause pipeline, and restart pipeline when queue bufferred enough data. Best RegardsZhao Liang From: Shenhong Wang [mailto:qc...@ho...] Sent: Wednesday, June 18, 2008 4:44 PMTo: Zhao Liang-E3423C; gst...@li...Subject: RE: [gst-embedded] Question on gst_plugin alsasink Hi, Zhao Liang:Generally, the aacdec &alsasink will not play out any audio frames(packets) after its source element has a break to send audio frames (packets) to them. It looks the alsasink drops all frames(packets) from the break. The break is needed because we have more video frames and sometime the wireless signal is not good. It looks the aacdec is slower than the expectation from alsasink.If so, how to fix the issue? thanks! best Regards!Shenhong Subject: RE: [gst-embedded] Question on gst_plugin alsasinkDate: Wed, 18 Jun 2008 14:29:27 +0800From: E3...@mo...To: qc...@ho...; gst...@li... Hi Shenhong, Your issue is very similar with the issue I even met. I think it is due to gstbaseaudiosink/gstaudiosink, it will drop the packets by gstringbuffer when read rate is bigger than write rate in ringbuffer, please see gstringbuffer.c gst_ring_buffer_commit_full (). For the rootcause, I think maybe the alsasink audiodevice buffer is too big or your aac decoder is too slow. Best RegardsZhao Liang From: gst...@li... [mailto:gst...@li...] On Behalf Of Shenhong WangSent: Wednesday, June 18, 2008 2:21 PMTo: gst...@li...Subject: [gst-embedded] Question on gst_plugin alsasink Dear all,Now we are using alsasink to play audio on Marvell PXA310 board. The audio is aac format. The audio frames(packets) are frequently sent to the aac decoder & alsasink to play out. Unfortunately only the begining frames can be played out and then nothing is played out. If we save those audio frames into a file, the aac decoder&alsasink can be successfully played out. It means the audio frames are ok. Could anyone tell me what's the difference for alsasink to process audio packets and files? How to fix the above issue? thank you very much! Best Regards!Shenhong WANG Connect to the next generation of MSN Messenger Get it now! Connect to the next generation of MSN Messenger Get it now! _________________________________________________________________ News, entertainment and everything you care about at Live.com. Get it now! http://www.live.com/getstarted.aspx |
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From: Zhao Liang-E. <E3...@mo...> - 2008-06-18 08:49:08
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Hi shenhong, A simply solution you can try. Put a queue before alsasink, when queue is dry, pause pipeline, and restart pipeline when queue bufferred enough data. Best Regards Zhao Liang ________________________________ From: Shenhong Wang [mailto:qc...@ho...] Sent: Wednesday, June 18, 2008 4:44 PM To: Zhao Liang-E3423C; gst...@li... Subject: RE: [gst-embedded] Question on gst_plugin alsasink Hi, Zhao Liang: Generally, the aacdec &alsasink will not play out any audio frames(packets) after its source element has a break to send audio frames (packets) to them. It looks the alsasink drops all frames(packets) from the break. The break is needed because we have more video frames and sometime the wireless signal is not good. It looks the aacdec is slower than the expectation from alsasink.If so, how to fix the issue? thanks! best Regards! Shenhong ________________________________ Subject: RE: [gst-embedded] Question on gst_plugin alsasink Date: Wed, 18 Jun 2008 14:29:27 +0800 From: E3...@mo... To: qc...@ho...; gst...@li... Hi Shenhong, Your issue is very similar with the issue I even met. I think it is due to gstbaseaudiosink/gstaudiosink, it will drop the packets by gstringbuffer when read rate is bigger than write rate in ringbuffer, please see gstringbuffer.c gst_ring_buffer_commit_full (). For the rootcause, I think maybe the alsasink audiodevice buffer is too big or your aac decoder is too slow. Best Regards Zhao Liang ________________________________ From: gst...@li... [mailto:gst...@li...] On Behalf Of Shenhong Wang Sent: Wednesday, June 18, 2008 2:21 PM To: gst...@li... Subject: [gst-embedded] Question on gst_plugin alsasink Dear all, Now we are using alsasink to play audio on Marvell PXA310 board. The audio is aac format. The audio frames(packets) are frequently sent to the aac decoder & alsasink to play out. Unfortunately only the begining frames can be played out and then nothing is played out. If we save those audio frames into a file, the aac decoder&alsasink can be successfully played out. It means the audio frames are ok. Could anyone tell me what's the difference for alsasink to process audio packets and files? How to fix the above issue? thank you very much! Best Regards! Shenhong WANG ________________________________ Connect to the next generation of MSN Messenger Get it now! <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&sou rce=wlmailtagline> ________________________________ Connect to the next generation of MSN Messenger Get it now! <http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&sou rce=wlmailtagline> |
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From: Shenhong W. <qc...@ho...> - 2008-06-18 08:45:08
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Brad, Yes, now I am using a live source via wireless signal. Questions: a) I don't know when the break will happen b) I don't know how to pause the pipeline How to move it forward? thanks! Best Regards! Shenhong WANG Subject: RE: [gst-embedded] Question on gst_plugin alsasinkDate: Wed, 18 Jun 2008 16:42:35 +0800From: bi...@mo...To: qc...@ho...; gst...@li... Why your source element has a break? do you use live source? I suggest you pause the pipleline at the moment. Brad From: Shenhong Wang [mailto:qc...@ho...] Sent: Wednesday, June 18, 2008 4:39 PMTo: Zhao Bin-E6223C; gst...@li...Subject: RE: [gst-embedded] Question on gst_plugin alsasink Brad,thanks! At the moment I didn't put any timestamp on the frames(buffer) yet.Generally, the aacdec &alsasink will not play out any audio frames(packets) after its source element has a break to send audio frames (packets) to them. It looks the alsasink drop all frames(packets) from the break. Why? How to fix it? Thanks! Best Regards!Shenhong WANG Subject: RE: [gst-embedded] Question on gst_plugin alsasinkDate: Wed, 18 Jun 2008 15:01:06 +0800From: bi...@mo...To: qc...@ho...; gst...@li... Hi, your issue seems that audio frame is delayed when arrivering alsasink. basesink will drop the delayed buffer and you could'nt hear any sound. Please check your timestamp of buffer. Brad From: gst...@li... [mailto:gst...@li...] On Behalf Of Shenhong WangSent: Wednesday, June 18, 2008 2:21 PMTo: gst...@li...Subject: [gst-embedded] Question on gst_plugin alsasink Dear all,Now we are using alsasink to play audio on Marvell PXA310 board. The audio is aac format. The audio frames(packets) are frequently sent to the aac decoder & alsasink to play out. Unfortunately only the begining frames can be played out and then nothing is played out. If we save those audio frames into a file, the aac decoder&alsasink can be successfully played out. It means the audio frames are ok. Could anyone tell me what's the difference for alsasink to process audio packets and files? How to fix the above issue? thank you very much! Best Regards!Shenhong WANG Connect to the next generation of MSN Messenger Get it now! Explore the seven wonders of the world Learn more! _________________________________________________________________ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE |
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