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From: amit s. <ami...@gm...> - 2010-12-06 07:59:54
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Hi, I want to compile gstreamer library through android NDK. I am not getting the source of gstreamer for android platform. So, if any body knows how to compile gstreamer with NDK please provide ur valuable input. Thanks On Fri, Dec 3, 2010 at 1:14 PM, Marco Balhis lesio <gib...@gm...>wrote: > Hi, > On Fri, Dec 3, 2010 at 7:55 AM, amit sharma <ami...@gm...> > wrote: > > Hello, > > > > I am new to android platform, and try to build gstreamer on android by > > using NDK but not getting any clue. I have already installed gstreamer on > > angstrom platform, but in android it seem some tedious work, so please > can > > any body tell me the link which i have to follow for building gstreamer > on > > android. > > Google is your friend:hankx > > http://www.elinux.org/images/a/a4/Android_and_Gstreamer.ppt > > thanks to Ubicast, there should even be a recording somewhere in the > web, but I could not find it from > http://gstconf.ubicast.tv/categories/conferences/ > > Probably the session was part of the CELF ELC conference, so the clip > should be somewhere else.. > > Regards > > > > > Thanks for your support. > > Regards > > Amit Sharma > > > > > ------------------------------------------------------------------------------ > > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > > Tap into the largest installed PC base & get more eyes on your game by > > optimizing for Intel(R) Graphics Technology. Get started today with the > > Intel(R) Software Partner Program. Five $500 cash prizes are up for > grabs. > > http://p.sf.net/sfu/intelisp-dev2dev > > _______________________________________________ > > Gstreamer-embedded mailing list > > Gst...@li... > > https://lists.sourceforge.net/lists/listinfo/gstreamer-embedded > |
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From: <Alb...@it...> - 2010-12-04 09:26:40
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Dear, I design a rtsp client player using gst-plugin-good, but I using "rtpdec" replace "rtpbin", my client could receive "sender report" from server, but could not send "receive report" to the server. I try using "rtpbin" , and it could do both receive/send actions. How could I send "receive report" to the server when I using "rtpdec" ? regards, Albert ==================================================================== 本信件可能包含工研院機密資訊，非指定之收件者，請勿使用或揭露本信件內容，並請銷毀此信件。 This email may contain confidential information. Please do not use or disclose it in any way and delete it if you are not the intended recipient. |
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From: Marco B. <gib...@gm...> - 2010-12-03 07:44:17
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Hi, On Fri, Dec 3, 2010 at 7:55 AM, amit sharma <ami...@gm...> wrote: > Hello, > > I am new to android platform, and try to build gstreamer on android by > using NDK but not getting any clue. I have already installed gstreamer on > angstrom platform, but in android it seem some tedious work, so please can > any body tell me the link which i have to follow for building gstreamer on > android. Google is your friend: http://www.elinux.org/images/a/a4/Android_and_Gstreamer.ppt thanks to Ubicast, there should even be a recording somewhere in the web, but I could not find it from http://gstconf.ubicast.tv/categories/conferences/ Probably the session was part of the CELF ELC conference, so the clip should be somewhere else.. Regards > > Thanks for your support. > Regards > Amit Sharma > > ------------------------------------------------------------------------------ > Increase Visibility of Your 3D Game App & Earn a Chance To Win $500! > Tap into the largest installed PC base & get more eyes on your game by > optimizing for Intel(R) Graphics Technology. Get started today with the > Intel(R) Software Partner Program. Five $500 cash prizes are up for grabs. > http://p.sf.net/sfu/intelisp-dev2dev > _______________________________________________ > Gstreamer-embedded mailing list > Gst...@li... > https://lists.sourceforge.net/lists/listinfo/gstreamer-embedded |
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From: amit s. <ami...@gm...> - 2010-12-03 05:55:52
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Hello, I am new to android platform, and try to build gstreamer on android by using NDK but not getting any clue. I have already installed gstreamer on angstrom platform, but in android it seem some tedious work, so please can any body tell me the link which i have to follow for building gstreamer on android. Thanks for your support. Regards Amit Sharma |
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From: DarkKnight B. <bin...@gm...> - 2010-11-08 12:24:36
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Hi This is just a reminder to my previous mail. Waiting for your response. Thanks praveen On Mon, Oct 25, 2010 at 8:01 PM, DarkKnight BrightWarrior < bin...@gm...> wrote: > Hi > We have to support different media format codecs in our linux machine. We > have to keep the size of distribution to minimum. > So I was searching for gstreamer codecs pack that supports various media > formats with minimum disk space. I saw one embedded device which supports > many formats and it was using fluendo codecs ( libgstflu...) combined with > codecs with name libgstmmrdemux.so, libgstmmrximagesink.so and > libgstmmrxvimagesink.so etc. I could not understand what are these > libgstmmr... codecs. Quick googling dis not show any Information. Does any > one know what are these codecs and origin package of these codecs. > > > thanks > praveen > |
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From: DarkKnight B. <bin...@gm...> - 2010-10-25 14:31:14
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Hi We have to support different media format codecs in our linux machine. We have to keep the size of distribution to minimum. So I was searching for gstreamer codecs pack that supports various media formats with minimum disk space. I saw one embedded device which supports many formats and it was using fluendo codecs ( libgstflu...) combined with codecs with name libgstmmrdemux.so, libgstmmrximagesink.so and libgstmmrxvimagesink.so etc. I could not understand what are these libgstmmr... codecs. Quick googling dis not show any Information. Does any one know what are these codecs and origin package of these codecs. thanks praveen |
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From: Stefan K. <en...@ho...> - 2010-10-15 19:42:49
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Am 28.09.2010 09:07, schrieb Jawad Akhtar: > Hi, > > I want to know does GStreamer provides video streaming functionality on SymbianOS ? GStreamer has only recently taken into use on Symbian OS. Right now imho it is used for the audiorecoders and a few audio apps on ovi store. Stefan > > Regards, > > Jawad. > > > > ------------------------------------------------------------------------------ > Start uncovering the many advantages of virtual appliances > and start using them to simplify application deployment and > accelerate your shift to cloud computing. > http://p.sf.net/sfu/novell-sfdev2dev > > > > _______________________________________________ > Gstreamer-embedded mailing list > Gst...@li... > https://lists.sourceforge.net/lists/listinfo/gstreamer-embedded |
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From: Felipe C. <fel...@gm...> - 2010-10-01 22:13:20
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Hi, It has been a long time but finally it's here; the first stable release of gst-openmax. Most likely it has been stable for quite some time, but only now the code-style has migrated to GStreamer one, and a bunch of patches from TI have been merged, so it seems like a good time to release. Also, this version now features a configuration mechanism, so you can select pretty much every combination of GStreamer elements you might need, and specify the OpenMAX IL library, component name, role, and even your desired rank. gst-openmax has been tested in multiple OpenMAX IL implementations, such as Bellagio, and Texas Instrument's OMAP. Depending on the implementation some changes might be needed, but ideally it should work straight out-of-the-box. If you find that some changes are needed, please contribute the changes back. Elements supported * MPEG-4 encoder/decoder * H.263 encoder/decoder * H.264 encoder/decoder * WMV decoder * Vorbis decoder * MP3 decoder * Volume filter Experimental elements (should work but not extensively tested) * AMR-NB encoder/decoder * AMR-WB encoder/decoder * AAC encoder/decoder * MP2 decoder * ADPCM encoder/decoder * G.711 encoder/decoder * G.729 encoder/decoder * iLBC encoder/decoder * JPEG encoder * video renderer * file reader Compared to previous pre-releases, this version now features a configuration mechanism, so you can select pretty much every combination of GStreamer elements you might need, and specify the OpenMAX IL library, component name, role, and even your desired rank. Tarballs can be found here: http://gstreamer.freedesktop.org/src/gst-openmax/ Compared to v0.10.0.5: Felipe Contreras (33): Update I420 to PackedPlanar util: trivial cleanup util: improve timeout messages util: cleanup ports when going to loaded util: increase timeout value base_videoenc: use 0 as automatic bitrate videodec: handle empty framerate build: update 'common' stuff util: fix compilation warning base_filter: trivial cleanup base_filter: more proper printf formatting base_sink: remove dead assignment videodec: fallback framerate to 0/1 tests: trivial fix Move component and library name fields to 'util' Reorganize core_new() Reorganize {library,component}-name assigns Make {component,library}-name read-only base: remove unused set_property() base: remove extra code plugin: reorganize code into get_config_path() plugin: add support for system-wide config Add example configuration plugin: reorganize element_table init plugin: make element_table const plugin: add support dependencies plugin: store element_table in plugin cache base-filter: improve EOS handling Fix compilation warnings Change to gst coding-style Generate ChangeLog build: remove patches extra dir build: add missing files to the dist Mark Nauwelaerts (1): base_filter: fix race condition when shutting down Rob Clark (21): Fixes for building for 64bit host Initial support for configuration file Add default configuration tests: update to use config file build: fixes for out-of-tree Use G_PARAM_STATIC_STRINGS basefilter: small fix for compile error Add GSTOMX_BOILERPLATE macros util: thread safety for _get_type() functions Construct GOmxPort objects in element constructor Simplify g_omx_port_setup() Don't hard-code port indexes core: call OMX_GetHandle in g_omx_core_new util: add some debug traces Add G_OMX_INIT_PARAM utility macro Add property helpers Add component-role support base: add input-buffers/output-buffers properties Add some debug traces Add GstOmxBaseAudioDec base class replace deprecated API Total contributions for v0.10.1: 1 David Schleef 2 Edward Hervey 370 Felipe Contreras 2 Frederik Vermelen 1 Frederik Vernelen 4 Jan Schmidt 1 Marco Ballesio 9 Mark Nauwelaerts 1 Olivier Crête 16 René Stadler 21 Rob Clark 4 Sebastian Dröge 1 Sriram Murthy 1 Stefan Kost 1 Tim-Philipp Müller Thanks to all the contributors! Cheers. -- Felipe Contreras |
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From: Jawad A. <jaw...@ya...> - 2010-09-28 06:07:21
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Hi,
I want to know does GStreamer provides video streaming functionality on
SymbianOS ?
Regards,
Jawad.
|
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From: Felipe C. <fel...@gm...> - 2010-09-20 11:27:05
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On Fri, Sep 17, 2010 at 2:49 AM, Feng Ye <fy...@br...> wrote: > I am working on a video dec plugin (based on hardware dec). I plan to use filesrc to read a raw H264 file and feed to my plugin. > The buffers from plugin's sink pad are the source data for decoding so they need to be DMA-able. Is there way to control how the buffers are allocated? Since they are allocated in filesrc so that's not possible? Looks like I will need to allocate my own buffer in my plugin and then copy them over? > > Also, a video frame may contain less bytes than what's in the buffer, what do I do with the remaining bytes? > >From the ffmpeg example (they are software based so the first dma question does not exist), they create a sub buffer of the remaining bytes and then does a buffer join. These are time consuming I think. I wonder if there are other faster solution? One thing I think might help is to have another plugin in between, which parse the file and only give one frame size of data to my plugin. But I am not sure if these kind of plugin exists. As Marco mentioned, you would wan to use h264parse for raw clips, but it's no actively maintained, so perhaps you would want to use a real demuxer instead. As of the question of DMA, we have the same issue in gst-dsp; currently GStreamer has no good control of DMA, so each buffer has to be mapped/unmapped constantly. If don't know of you have some sort of MMU so that you can do the same, but if not a memcpy is unavoidable, imagining a perfect scenario where filesrc is mmaping the file, somebody would need to copy that data to your buffer. Unless pull mode is used, which I don't know if it would work, nor if it makes sense. Cheers. -- Felipe Contreras |
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From: Felipe C. <fel...@gm...> - 2010-09-20 09:45:42
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Hi, gst-dsp is a GStreamer plug-in to utilize Texas Intruments' DSP algorithms for OMAP3 platforms using the tidspbridge driver. This is a major release, many features have been implemented: * Refactoring of DMA API * Fixes for newer DSP API versions * Add JPEG decoder implementation * Fix WMV support * Reorganize the way EOS is handled * Improve H.264 keyframe handling * Improve deinitialization on fatal errors (like MMU faults) * Fixes for ARMv7 speculative pre-fetching * Several improvements for encoding bitrate calculation * Add support for I420 color format * Correctly propagate aspect-ratio And the usual bunch of cleanups and reorganization. Also it's worth noting that this code has been tested on Nokia N900, beagleboard, and future Nokia hw; with different DSP binaries; L23.i3.8, L23.i3.3, and 0.3.5; and different DSP bridge driver versions. For very old binaries, build with DSP_API=0, it's not necessary to set SN_API yet, as the structures are binary compatible with all the versions. Moreover, this time I decided to package, this, and some related packages in tarballs in order to help some distributions: http://code.google.com/p/gst-dsp/downloads This time we have slightly more contributors, including even some patches from TI. Thanks everyone! Unfortunately this is still not ready for 1.0, the parsers need to be cleaned up, fixed, and merged. And once tidspbridge 0.4 is tagged, move to the new DMA API ioctls. Finally, most likely the core of gst-dsp will move to a separate library (not related to GStreamer): libtidsp, in order to be useful in other projects. A sample of that has been submitted to FFmpeg for review: http://article.gmane.org/gmane.comp.video.ffmpeg.devel/116798 Enjoy ;) Arvind Gupta (3): vdec: MPEG4 sn to use double buffer hack: venc: strip SPS and PPS headers on keyframes vdec: return last frame EOS from MPEG4, H.263 and WMV decoders Felipe Contreras (111): venc: fix default mode base: reorganize status check in output_loop() base: reorganize error checks in output_loop() log: fix log level 3 when DEBUG is on base: continue deinit on errors base: make sure stop message was sent base: detect algorithm errors base: don't print messages as errors base: reorganize error handling a bit base: exit dsp_thread more cleanly Allow more granularity in dspbridge API build: add DSP_API option Plug a few memleaks when node_create() fails bridge: trivial cleanups bridge: fix a small exceptional leak vdec: always clear params build: add option for socket-node API version Trivial cleanups dmm-buffer: we are using a proc handle, not node base: avoid yoda conditions base: improve send_buffer() venc: remove unnecessary buffer_clean()s dummy: invalidate before sending vdec: remove h264dec_in_stream_params Trivial cleanups base: check for errors in dsp_wait_for_events() base: avoid buffer duplication on corner-case Refactor some code into dmm_buffer_calloc() Refactor code into gstdsp_port_setup_params() Add new gstdsp_send_alg_ctrl() base: free alg_ctrl on exceptions Rework cache maintainance functions base: always invalidate out buffers afterwards Add DMA direction information Start using new DMA API dmm-buffer: remove _unmap() dmm-buffer: we know the page size dmm-buffer: let's not be too smart about unreserve dmm-buffer: put map and reserve together Manually call dmm_buffer_map() Add manual calls to dmm_buffer_unmap() dmm-buffer: remove clean/invalidate/flush build: general improvements venc: fix for MPEG-4 SN_API=1 venc: add keyframe-interval property Move ARRAY_SIZE to util.h vdec: log actual SN used Trivial improvements venc: improve bitrate calculation vdec: propagate aspect-ratio Support multiple SN_API's venc: register conversions library for jpegenc venc: fix jpegenc params Remove unnecessary assigns build: add dist target bridge: fix bad ioctls bridge: update copyright log: generic improvements vdec: fix trivial warning build: check for deprecated gst stuff Fix depreated stuff from gst vdec: trivial cleanup Create ports in base class log: trivial whitespace cleanups Update copyright notices Update licence notices Add LICENSE file build: only link to needed libraries get-version: trivial improvements build: a bit more warnings parse: fix compilation warning base: don't loose buffers on EOS base: fix a non-atomic check for self->status base: use atomic 'deferred_eos' instead of 'eos' base: reorganize EOS handling base: remove redundant checks for use_eos_align base: clear deferred_eos when appropriate base: refactor code into pause_task() Remove 'use_eos_align' base: a few more compiler hints venc: extract codec-data for MPEG-4 venc: simplify avr mode workaround venc: trivial cleanups hack: h264enc: manually force keyframes venc: add property for intra-refresh venc: fix check_supported_levels() venc: pick max bitrate at same level of mbps venc: limit max_bitrate to the true maximum venc: add check for invalid level venc: cap bitrate to max_bitrate venc: remove redundant max_bitrate check for H.264 vdec: calculate real coded framesize vdec: add crop framesize vdec: update buffer len based on crop size base: update gst buffersize based on dmm_buffer len Rename dsp_{start,stop} dmm_buffer: whitespace cleanups bridge: whitespace cleanups bridge: trivial cleanups bridge: reorganize return code handling bridge: cleanup return codes bridge: remove typedefs Remove usage of typedefs bridge: truly remove typedefs dummy: use correct socket-node dummy: fix mapping sequence dummy: improve mapping Trivial cleanups dummy: add proper DSP event handling dummy: improve error handling Update README Juha Alanen (7): vdec: update WMV create phase parameters base: use hexadecimal format when printing the error code vdec: register conversions library vdec: add send_cb for MPEG4 decoder vdec: add recv_cb for MPEG4 decoder vdec: enable {send,recv}_cb for H.263 decoder venc: fix supported levels for H.264 Karunesha Uh (2): vdec: add support for I420 base: add virtual method flush_frames Laxman Raju (1): venc: set right profile for JPEG enc Marco Ballesio (1): hack: venc: avoid intra frame interval Mark Nauwelaerts (16): base: appropriately flush queued events base: keyframe marking must be thread-safe base: additional checks for NULL buffers base: ensure codec data is sent to node and not silently dropped base: enhance eos aligned EOS sending venc: add support for I420 YUV format base: disable queues when pausing src task base: free comm buffers in dsp_stop base: do not consume buffer timestamp for a 0-length output buffer venc: adjust h264 encoder frametype check base: more state variable cleanup when stopping base: add reset() vmethod venc: implement reset() vmethod for state variable cleanup vdec: fix codec-data transformation venc: update jpegenc params to release L23.i3.4 venc: no automagic changes to property values Martin Storsjo (1): bridge: fix for DSP_API=0 Miguel Verdu (2): venc: add supported levels venc: H.264 CBR mode uses CIR Olivier Crête (1): venc: add max-bitrate property Raju Laxman (1): venc: update jpeg encoder API Tommi Myöhänen (1): base: check EOS before reallocating out buffer Víctor Manuel Jáquez Leal (23): bridge: add dsp_node_connect() bridge: add dsp_stream_open() bridge: add dsp_stream_close() bridge: add dsp_stream_allocate_buffers() bridge: add dsp_stream_idle() bridge: add dsp_stream_get_info() bridge: add dsp_stream_reclaim() bridge: add dsp_stream_free_buffers() bridge: add dsp_stream_issue() dsp-bridge: check for attr before deferring base: add du_port_alloc_buffers() base: delay the comm port allocation Delay allocation of port buffers jpegdec: initial implementation bridge: dsp_uuid_t normally is const vdec: add and use configure_caps() vdec: i420 support for jpgdec vdec: check for structures in caps vdec: allowed_caps makes more sense vdec: trivial cleanup bridge: add dsp_stop() bridge: add dsp_load() bridge: add dsp_start() -- Felipe Contreras |
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From: Feng Y. <fy...@br...> - 2010-09-16 23:49:45
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Hello there, I am working on a video dec plugin (based on hardware dec). I plan to use filesrc to read a raw H264 file and feed to my plugin. The buffers from plugin's sink pad are the source data for decoding so they need to be DMA-able. Is there way to control how the buffers are allocated? Since they are allocated in filesrc so that's not possible? Looks like I will need to allocate my own buffer in my plugin and then copy them over? Also, a video frame may contain less bytes than what's in the buffer, what do I do with the remaining bytes? >From the ffmpeg example (they are software based so the first dma question does not exist), they create a sub buffer of the remaining bytes and then does a buffer join. These are time consuming I think. I wonder if there are other faster solution? One thing I think might help is to have another plugin in between, which parse the file and only give one frame size of data to my plugin. But I am not sure if these kind of plugin exists. Thanks, Feng |
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From: Michael S. <ms...@xi...> - 2010-09-14 04:49:42
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Basically only when GST_BOILERPLATE/GST_BOILERPLATE_FULL doesn't work for your purposes. Mike On Mon, Sep 13, 2010 at 9:42 PM, Feng Ye <fy...@br...> wrote: > Thanks Mike, that explains why I can't find this function in certain cases. > Is there a general guide line as to when I should implement my own get_type() function? > > > Feng > > -----Original Message----- > From: mlr...@gm... [mailto:mlr...@gm...] On Behalf Of Michael Smith > Sent: Monday, September 13, 2010 5:08 PM > To: Feng Ye > Cc: gst...@li... > Subject: Re: [gst-embedded] what is the GstType that's passed to the gst_element_register() call for plugin_init? > > On Mon, Sep 13, 2010 at 4:51 PM, Feng Ye <fy...@br...> wrote: >> Hello there, >> >> >> >> I am doing my plugin and I have a question regarding the last parameter in >> the gst_element_register() call, which is the GstType. >> >> In the base plugin package, I see normally it's defined in the header file >> to a function, but for some plugins this function is there in the package, >> for others it's not. > > In most plugins, the get_type() function is provided by the > GST_BOILERPLATE or GST_BOILERPLATE_FULL macros. > > Mike > > > |
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From: Feng Y. <fy...@br...> - 2010-09-14 04:42:34
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Thanks Mike, that explains why I can't find this function in certain cases. Is there a general guide line as to when I should implement my own get_type() function? Feng -----Original Message----- From: mlr...@gm... [mailto:mlr...@gm...] On Behalf Of Michael Smith Sent: Monday, September 13, 2010 5:08 PM To: Feng Ye Cc: gst...@li... Subject: Re: [gst-embedded] what is the GstType that's passed to the gst_element_register() call for plugin_init? On Mon, Sep 13, 2010 at 4:51 PM, Feng Ye <fy...@br...> wrote: > Hello there, > > > > I am doing my plugin and I have a question regarding the last parameter in > the gst_element_register() call, which is the GstType. > > In the base plugin package, I see normally it's defined in the header file > to a function, but for some plugins this function is there in the package, > for others it's not. In most plugins, the get_type() function is provided by the GST_BOILERPLATE or GST_BOILERPLATE_FULL macros. Mike |
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From: Michael S. <ms...@xi...> - 2010-09-14 00:07:41
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On Mon, Sep 13, 2010 at 4:51 PM, Feng Ye <fy...@br...> wrote: > Hello there, > > > > I am doing my plugin and I have a question regarding the last parameter in > the gst_element_register() call, which is the GstType. > > In the base plugin package, I see normally it’s defined in the header file > to a function, but for some plugins this function is there in the package, > for others it’s not. In most plugins, the get_type() function is provided by the GST_BOILERPLATE or GST_BOILERPLATE_FULL macros. Mike |
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From: Feng Y. <fy...@br...> - 2010-09-13 23:51:47
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Hello there, I am doing my plugin and I have a question regarding the last parameter in the gst_element_register() call, which is the GstType. In the base plugin package, I see normally it's defined in the header file to a function, but for some plugins this function is there in the package, for others it's not. For example, gst/gdp/gstgdpdepay.c used GST_TYPE_GDP_DEPAY: gst_element_register (plugin, "gdpdepay", GST_RANK_NONE, GST_TYPE_GDP_DEPAY) And gstdgpdepay.h has this define: #define GST_TYPE_GDP_DEPAY (gst_gdp_depay_get_type()) However I can't find this gst_gdp_depay_get_type() function implementation in the whole package. Many of other plugins have the same thing. One example on the other case is in gst/tcp/gsttcpplugin.c, which used type GST_TYPE_TCP_CLIENT_SINK, and this can be traced to function gst_tcp_client_sink_get_type(), which is implemented in gsttcpclientsink.c. So I wonder which is the way I should go, is the unimplemented function hidden somewhere? Why it won't cause compile problem? Thanks, Feng |
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From: Andrey N. <and...@ya...> - 2010-09-13 08:33:51
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Hi Fabio, >I am working on a ARM video server streaming a H264 flow of data over RTP >protocol. >I am using appsrc module to inject data into the pipeline. >The server must start/stop streaming on remote-request basis; I am working on the similar functionality for my hobby project (www.gitorious.org/veter/pages/Home) and were also facing different problems with restarting pipeline. That is why I decide to completely destroy/create pipeline instead of pausing it. It does not take much time to (re)create the pipeline and this delay is not really a problem for the interactive users. Sorry for not answering your question directly, but I thought that the idea could be used at least as a workaround in your case. Regards, Andrey. |
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From: f1962 <f1...@us...> - 2010-09-09 20:08:38
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Hi all... I am working on a ARM video server streaming a H264 flow of data over RTP protocol. I am using appsrc module to inject data into the pipeline. The server must start/stop streaming on remote-request basis; to satisfy this request I do gst_element_set_state (pipeline, GST_STATE_PLAYING) to start the streaming and gst_element_set_state (pipeline, GST_STATE_PAUSED) to stop the streaming. In other words the operation sequence is: - create the pipeline and set its state to GST_STATE_PAUSED; - wait for a StartStreaming request; - set the pipeline status to GST_STATE_PLAYING and begin to inject data into the pipeline by means of push-buffer signal; - when a StopStreaming request arrive, set GST_STATE_PAUSED the pipeline.... - and so on... The problem is that after the second StartStreaming request is received , all the push-buffer operations return with GST_FLOW_UNEXPECTED error. Can someone can help me? Thanks Fabio |
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From: Wim T. <wim...@gm...> - 2010-08-26 11:53:16
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On Thu, 2010-08-26 at 13:41 +0200, Alberto Milla wrote: > Hi, I am working with an at91sam9g45ekes and openembedded at its stable branch. > > I tried to do a simple hello world program with gstreamer but it did > not work. After that I started testing gstreamer with the command: > > # gst-launch audiotestsrc ! audioconvert ! alsasink > > I get: > > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > Pipeline is PREROLLED ... > Setting pipeline to PLAYING ... > New clock: GstAudioSinkClock > > And gst-launch get stucked without playing any buzzer sound and to > exit I had to do a ctrl-c and after that gstreamer stopped in: Your alsasink seems broken, it should play sound. Maybe the volume it too low? > > Caught interrupt -- handling interrupt. > Interrupt: Stopping pipeline ... > Execution ended after 11443136693 ns. > Setting pipeline to PAUSED ... > > After that, I have to do a reset to the board. But if I play a OGG file: Sounds like a driver problem to me. > > # gst-launch filesrc location=music.ogg ! oggdemux ! vorbisdec ! alsasink > > I get only a warning but no sound: > > WARNING: erroneous pipeline: could not link vorbisdec0 to alsasink0 You need to add audioconvert ! audioresample in front of alsasink. > > And to finish, I tried too with a mp3 file and I had the audiotestsrc problem: > > # gst-launch filesrc location=music.mp3 ! mad ! alsasink > > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > Pipeline is PREROLLED ... > Setting pipeline to PLAYING ... > New clock: GstAudioSinkClock > Again, this should play the sound, if not, check the volume and the driver. Wim > > Can you help me? > > > Thanks, > > Alberto > > ------------------------------------------------------------------------------ > Sell apps to millions through the Intel(R) Atom(Tm) Developer Program > Be part of this innovative community and reach millions of netbook users > worldwide. Take advantage of special opportunities to increase revenue and > speed time-to-market. Join now, and jumpstart your future. > http://p.sf.net/sfu/intel-atom-d2d > _______________________________________________ > Gstreamer-embedded mailing list > Gst...@li... > https://lists.sourceforge.net/lists/listinfo/gstreamer-embedded |
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From: Alberto M. <am...@vi...> - 2010-08-26 11:42:28
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Hi, I am working with an at91sam9g45ekes and openembedded at its stable branch. I tried to do a simple hello world program with gstreamer but it did not work. After that I started testing gstreamer with the command: # gst-launch audiotestsrc ! audioconvert ! alsasink I get: Setting pipeline to PAUSED ... Pipeline is PREROLLING ... Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstAudioSinkClock And gst-launch get stucked without playing any buzzer sound and to exit I had to do a ctrl-c and after that gstreamer stopped in: Caught interrupt -- handling interrupt. Interrupt: Stopping pipeline ... Execution ended after 11443136693 ns. Setting pipeline to PAUSED ... After that, I have to do a reset to the board. But if I play a OGG file: # gst-launch filesrc location=music.ogg ! oggdemux ! vorbisdec ! alsasink I get only a warning but no sound: WARNING: erroneous pipeline: could not link vorbisdec0 to alsasink0 And to finish, I tried too with a mp3 file and I had the audiotestsrc problem: # gst-launch filesrc location=music.mp3 ! mad ! alsasink Setting pipeline to PAUSED ... Pipeline is PREROLLING ... Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstAudioSinkClock Can you help me? Thanks, Alberto |
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From: Stefan K. <en...@ho...> - 2010-08-25 20:05:43
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Am 02.08.2010 11:04, schrieb Markus Schulte:
> hi,
>
> i spent my weekend trying to xcompile gstreamers upnp interface
> gmediarender for openwrt.
GMediarenderer is a project independent of gstreamer. A more up-to-date upnp
solution might be GUPnP.
>
> i have Makefile i am proud of!
>
> well, it compiles but does not run.
>
> while gst-inspect gives me proper output and i can play music with
> gst-launch, the libraries apparently don't find any plugin when linked
> with gmediarender. at least due to my understanding. the problem may be
> somewhere else.
>
> compiling and linking the exact same sources for ubuntu works great.
>
> this is what i get from gmediarender:
>
> ----
> root@OpenWrt:~# gmediarender --gst-plugin-load=alsa,udp,mad -I 192.168.200.2
> Registering support for 'audio/mpeg'
> Registering support for 'audio/x-mpeg'
> Registering support for 'audio/x-raw-int'
> Registering support for 'audio/x-iec958'
>
> (gmediarender:25351): GStreamer-CRITICAL **: gst_element_get_bus:
> assertion `GST_IS_ELEMENT (element)' failed
>
> (gmediarender:25351): GStreamer-CRITICAL **: gst_bus_add_watch_full:
> assertion `GST_IS_BUS (bus)' failed
>
> (gmediarender:25351): GStreamer-CRITICAL **: gst_object_unref: assertion
> `object != NULL' failed
>
> (gmediarender:25351): GStreamer-CRITICAL **: gst_element_set_state:
> assertion `GST_IS_ELEMENT (element)' failed
> Error: pipeline doesn't want to get ready
> registering '/upnp/rendertransportSCPD.xml'
> registering '/upnp/renderconnmgrSCPD.xml'
> registering '/upnp/rendercontrolSCPD.xml'
> Ready for rendering..
> ----
>
> it lies! it definetly is not ready for rendering. the status line is
> missing too ('GStreamer: play: State change: OLD: 'NULL', NEW: 'READY',
> PENDING: 'VOID_PENDING'
> ').
I'd suggest to file a bug against gmediarender.
>
> this is what i get from gst-inspect:
> ----
> root@OpenWrt:~# gst-inspect
> coreelements: multiqueue: MultiQueue
> coreelements: typefind: TypeFind
> coreelements: tee: Tee pipe fitting
> coreelements: filesink: File Sink
> coreelements: queue2: Queue 2
> coreelements: queue: Queue
> coreelements: identity: Identity
> coreelements: filesrc: File Source
> coreelements: fdsink: Filedescriptor Sink
> coreelements: fdsrc: Filedescriptor Source
> coreelements: fakesink: Fake Sink
> coreelements: fakesrc: Fake Source
> coreelements: capsfilter: CapsFilter
> coreindexers: memindex: A index that stores entries in memory
> udp: udpsrc: UDP packet receiver
> udp: dynudpsink: UDP packet sender
> udp: multiudpsink: UDP packet sender
> udp: udpsink: UDP packet sender
> mad: mad: mad mp3 decoder
> alsa: alsasink: Audio sink (ALSA)
> alsa: alsasrc: Audio source (ALSA)
> alsa: alsamixer: Alsa mixer
> staticelements: bin: Generic bin
> staticelements: pipeline: Pipeline object
This is core and a few of base. Not much. Seems to be tailored to mp3 playback.
> ----
>
> "gmediarender -I 192.168.200.2 --gst-plugin-load=udp,mad
> --gstout-audiosink=alsasink" segfaults :-(
>
> "strace gmediarender -I 192.168.200.2 --gst-plugin-load=udp,mad
> --gstout-audiosink=alsasink 2>/tmp/strace.log"
>
> shows that, while it seems to properly find the plugins:
>
> open("/usr/lib/gstreamer-0.10/libgstudp.so", O_RDONLY) = 3
> open("/usr/lib/libmad.so.0", O_RDONLY) = 3
> open("/usr/lib/gstreamer-0.10/libgstalsa.so", O_RDONLY) = 3
>
> it still segfaults.
Yes, those assertions need to be fixed for sure.
Stefan
>
> this is a tail on strace.log:
> ---
> write(2, "\n(gmediarender:25649): GStreamer"..., 98
> (gmediarender:25649): GStreamer-CRITICAL **: gst_object_unref: assertion
> `object != NULL' failed
> ) = 98
> write(1, "Setting audio sink to alsasink\n", 31) = 31
> socketpair(PF_FILE, SOCK_STREAM, 0, [8, 9]) = 0
> fcntl64(8, F_SETFL, O_RDONLY|O_NONBLOCK) = 0
> fcntl64(9, F_SETFL, O_RDONLY|O_NONBLOCK) = 0
> --- SIGSEGV (Segmentation fault) @ 0 (0) ---
> +++ killed by SIGSEGV +++
> Segmentation fault
> ---
>
> to make the picture complete, this is the linker command:
>
> ---
> mipsel-openwrt-linux-uclibc-gcc -D_GNU_SOURCE -Os -pipe -mips32
> -mtune=mips32 -funit-at-a-time -fhonour-copts -msoft-float -fpic -Wall
> -I/home/markus/hotspot/openwrt/backfire/staging_dir/target-mipsel_uClibc-0.9.30.1/usr/include
> -I/home/markus/hotspot/openwrt/backfire/staging_dir/target-mipsel_uClibc-0.9.30.1/include
> -I/home/markus/hotspot/openwrt/backfire/staging_dir/toolchain-mipsel_gcc-4.3.3+cs_uClibc-0.9.30.1/usr/include
> -I/home/markus/hotspot/openwrt/backfire/staging_dir/toolchain-mipsel_gcc-4.3.3+cs_uClibc-0.9.30.1/include
> -L/home/markus/hotspot/openwrt/backfire/staging_dir/target-mipsel_uClibc-0.9.30.1/usr/lib
> -L/home/markus/hotspot/openwrt/backfire/staging_dir/target-mipsel_uClibc-0.9.30.1/lib
> -L/home/markus/hotspot/openwrt/backfire/staging_dir/toolchain-mipsel_gcc-4.3.3+cs_uClibc-0.9.30.1/usr/lib
> -L/home/markus/hotspot/openwrt/backfire/staging_dir/toolchain-mipsel_gcc-4.3.3+cs_uClibc-0.9.30.1/lib
> -L/home/markus/hotspot/openwrt/backfire/staging_dir/target-mipsel_uClibc-0.9.30.1/usr/lib/libintl/lib
> -L/home/markus/hotspot/openwrt/backfire/staging_dir/target-mipsel_uClibc-0.9.30.1/usr/lib/libiconv/lib
> -Wl,-rpath-link=/home/markus/hotspot/openwrt/backfire/staging_dir/target-mipsel_uClibc-0.9.30.1/usr/lib
> -L/home/markus/hotspot/openwrt/backfire/staging_dir/target-mipsel_uClibc-0.9.30.1/usr/lib
> -L/home/markus/hotspot/openwrt/backfire/staging_dir/target-mipsel_uClibc-0.9.30.1/usr/lib
> -L/home/markus/hotspot/openwrt/backfire/staging_dir/target-mipsel_uClibc-0.9.30.1/lib
> -L/home/markus/hotspot/openwrt/backfire/staging_dir/toolchain-mipsel_gcc-4.3.3+cs_uClibc-0.9.30.1/usr/lib
> -L/home/markus/hotspot/openwrt/backfire/staging_dir/toolchain-mipsel_gcc-4.3.3+cs_uClibc-0.9.30.1/lib
> -L/home/markus/hotspot/openwrt/backfire/staging_dir/target-mipsel_uClibc-0.9.30.1/usr/lib/libintl/lib
> -L/home/markus/hotspot/openwrt/backfire/staging_dir/target-mipsel_uClibc-0.9.30.1/usr/lib/libiconv/lib
> -Wl,-rpath-link=/home/markus/hotspot/openwrt/backfire/staging_dir/target-mipsel_uClibc-0.9.30.1/usr/lib
> -o gmediarender main.o upnp.o upnp_control.o upnp_connmgr.o
> upnp_transport.o upnp_device.o upnp_renderer.o webserver.o
> output_gstreamer.o xmlescape.o -pthread
> -L/home/markus/hotspot/openwrt/backfire/staging_dir/target-mipsel_uClibc-0.9.30.1/usr//lib
> -lgstreamer-0.10 -lgobject-2.0 -lgmodule-2.0 -lgthread-2.0 -lxml2
> -lglib-2.0 -lintl -liconv -lupnp
>
> ---
>
>
> for the sake of my next weekend and all those who wait for an openwrt
> upn media renderer, please help,
>
>
> cheers,
>
> /markus
>
>
>
>
>
> ------------------------------------------------------------------------------
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> Plug-In Development Kit to bring their C/C++ apps to Palm for a share
> of $1 Million in cash or HP Products. Visit us here for more details:
> http://p.sf.net/sfu/dev2dev-palm
>
>
>
> _______________________________________________
> Gstreamer-embedded mailing list
> Gst...@li...
> https://lists.sourceforge.net/lists/listinfo/gstreamer-embedded
|
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From: Edward H. <edw...@co...> - 2010-08-25 08:56:28
|
Hi, On Tue, 2010-08-24 at 18:26 -0700, Feng Ye wrote: > Mike, > > Thanks so much! This solves the problem. > This option was not there when I do "configure --help". > > Now I have lots of undefined references like these, I think most likely libglib is not up to date: > > .../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_param_spec_object' > .../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_utf8_validate' > .../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_date_get_month' > .../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_value_get_float' > .../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_memdup' > .../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_type_interface_peek' > .../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_value_set_int64' > .../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_thread_pool_new' > .../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_flags_get_first_value' > .../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_slist_remove' > .../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_mkdir_with_parents' > .../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `glib_major_version' > .../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_value_unset' > .../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_source_set_priority' > .../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_strdupv' Most of those symbols have been present in glib since the dawn of time. Looks like it can't find libglib. run "make V=1" and make sure -lglib-2.0 is being used at the compilation/linking stages. Edward > > > > Feng > > > > ------------------------------------------------------------------------------ > Sell apps to millions through the Intel(R) Atom(Tm) Developer Program > Be part of this innovative community and reach millions of netbook users > worldwide. Take advantage of special opportunities to increase revenue and > speed time-to-market. Join now, and jumpstart your future. > http://p.sf.net/sfu/intel-atom-d2d > _______________________________________________ > Gstreamer-embedded mailing list > Gst...@li... > https://lists.sourceforge.net/lists/listinfo/gstreamer-embedded -- Edward Hervey -- Collabora Multimedia Lead Platforms Engineer Co-Founder |
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From: Feng Y. <fy...@br...> - 2010-08-25 01:27:08
|
Mike, Thanks so much! This solves the problem. This option was not there when I do "configure --help". Now I have lots of undefined references like these, I think most likely libglib is not up to date: ../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_param_spec_object' ../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_utf8_validate' ../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_date_get_month' ../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_value_get_float' ../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_memdup' ../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_type_interface_peek' ../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_value_set_int64' ../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_thread_pool_new' ../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_flags_get_first_value' ../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_slist_remove' ../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_mkdir_with_parents' ../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `glib_major_version' ../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_value_unset' ../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_source_set_priority' ../../../gst/.libs/libgstreamer-0.10.so: undefined reference to `g_strdupv' Feng |
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From: Michael S. <ms...@xi...> - 2010-08-25 00:08:39
|
On Tue, Aug 24, 2010 at 4:41 PM, Feng Ye <fy...@br...> wrote: > Hello there, > > > > I am porting gstreamer to our ARM based system. I had many problems during > the process (mainly with libglib) and now my problem is this > INTROSPECTION_SCANNER. I don't directly know what the solution to your problem is. However, it's very unlikely that you actually want to use the gobject introspection stuff, so configuring with --disable-introspection will fix this issue. Mike |
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From: Feng Y. <fy...@br...> - 2010-08-24 23:42:13
|
Hello there, I am porting gstreamer to our ARM based system. I had many problems during the process (mainly with libglib) and now my problem is this INTROSPECTION_SCANNER. I checked my config.log and I see this is not set. The package I tried to port is 0.10.30, the error message is: ... CC libgstreamer_0.10_la-gstparse.lo CC libgstreamer_0.10_la-gstregistrybinary.lo CC libgstreamer_0.10_la-gstenumtypes.lo CC libgstreamer_0.10_la-gstmarshal.lo CCLD libgstreamer-0.10.la GEN Gst-0.10.gir /bin/sh: -v: command not found make[6]: *** [Gst-0.10.gir] Error 127 make[5]: *** [all-recursive] Error 1 make[4]: *** [all] Error 2 make[3]: *** [all-recursive] Error 1 make[2]: *** [all] Error 2 I wonder what this should set to? Thanks, Feng |
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