From boning.zhang at intel.com Tue Sep 1 08:12:14 2009 From: boning.zhang at intel.com (Zhang, Boning) Date: Tue, 1 Sep 2009 14:12:14 +0800 Subject: [gst-devel] faad streaming decode problem Message-ID: <037F493892196B458CD3E193E8EBAD4F01EC6D1974@pdsmsx502.ccr.corp.intel.com> Sending: gst-launch audiotestsrc ! audioconvert ! audioresample ! faac ! rtpmp4apay ! udpsink host=localhost port=5001 Receiving : gst-launch udpsrc port=5001 caps="application/x-rtp,media=(string)audio,clock-rate=(int)44100,payload=(int)96,encoding-name=(string)MP4A-LATM" ! rtpmp4adepay ! faad ! filesink location=./abc If I run this receiving cmd, it reports no error but the size of file "abc" is 0, but if I delete the faad use "...! rtpmp4adepay !filesink location=./abc" I can get a file without 0 size.And if I replace the faad with ffdec_aac it will not decode with the error "Could not decode stream" Thanks ------------------------------------ Boning,Zhang -------------- next part -------------- An HTML attachment was scrubbed... URL: From miquel.farre at gmail.com Tue Sep 1 12:55:15 2009 From: miquel.farre at gmail.com (=?ISO-8859-1?Q?Miquel_=C0ngel_Farr=E9?=) Date: Tue, 1 Sep 2009 12:55:15 +0200 Subject: [gst-devel] loosing sync mpegtsdemux Message-ID: <49c4aa740909010355n6a0e5188t4ad503666e232629@mail.gmail.com> Hi, I am working on an application that starts with this pipeline: filesrc location = filename ! mpegtsparse ! mpegtsdemux when mpegtsdemux have a new pad, it calls the function on_new_pad: in case of video it completes the pipeline with: queue ! mpeg2dec ! videoscale ! capsfilter ! glupload ! fakesink (with sync=true) in case of dsmcc content: queue ! appsink The problem is that the video part plays faster than the usual, nevertheless if I comment all the dsmcc part on "on_pad_added" function, video part plays correctly. I can't found where is the problem, but I think it is related with the dsmcc part pipeline, here you can found the link to "on_pad_added" function: http://pastebin.ca/1550152 Thanks, Miquel -------------- next part -------------- An HTML attachment was scrubbed... URL: From trungnthut at gmail.com Tue Sep 1 12:59:05 2009 From: trungnthut at gmail.com (=?UTF-8?B?VGjDoG5oIFRydW5nIE5ndXnhu4Vu?=) Date: Tue, 1 Sep 2009 17:59:05 +0700 Subject: [gst-devel] _chain() function for GstBin subclass In-Reply-To: <5b9896820908310355j221c06edw84c6252fc7439969@mail.gmail.com> References: <5b9896820908282235k7f6e6c1cl943f34d66614c42e@mail.gmail.com> <1251539630.4740.1.camel@zingle> <5b9896820908310355j221c06edw84c6252fc7439969@mail.gmail.com> Message-ID: <5b9896820909010359j5b0c5bf9p25c4c2ffad494569@mail.gmail.com> Look like I did something wrong. I restarted with some fresh code and it work now ! Thanks 2009/8/31 Th?nh Trung Nguy?n > So I wonder why there's nothing come to videosink. I've already linked > ghost pads to needed elements. > > > On Sat, Aug 29, 2009 at 4:53 PM, Tim-Philipp M?ller wrote: > >> On Sat, 2009-08-29 at 12:35 +0700, Th?nh Trung Nguy?n wrote: >> >> > So, what do I need to do when inherit GstBin, mean: which methods are >> > needed to have. What's replaced for _chain() function ? >> >> There is no replacement. Data is routed via ghostpads to/from elements >> inside your bin. >> >> Cheers >> -Tim >> >> >> >> >> ------------------------------------------------------------------------------ >> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 >> 30-Day >> trial. Simplify your report design, integration and deployment - and focus >> on >> what you do best, core application coding. Discover what's new with >> Crystal Reports now. http://p.sf.net/sfu/bobj-july >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > > > -- > Cheers ! > > trungnt > -- Cheers ! trungnt -------------- next part -------------- An HTML attachment was scrubbed... URL: From miquel.farre at gmail.com Tue Sep 1 14:04:30 2009 From: miquel.farre at gmail.com (=?ISO-8859-1?Q?Miquel_=C0ngel_Farr=E9?=) Date: Tue, 1 Sep 2009 14:04:30 +0200 Subject: [gst-devel] loosing sync mpegtsdemux In-Reply-To: <49c4aa740909010355n6a0e5188t4ad503666e232629@mail.gmail.com> References: <49c4aa740909010355n6a0e5188t4ad503666e232629@mail.gmail.com> Message-ID: <49c4aa740909010504q56dafe19ue5eb8ccf3ec04e3b@mail.gmail.com> It seems solved just adding the property of async = true on all appsink elements of dsmcc part.. El 1 / setembre / 2009 12:55, Miquel ?ngel Farr? ha escrit: > Hi, > > I am working on an application that starts with this pipeline: > > filesrc location = filename ! mpegtsparse ! mpegtsdemux > > when mpegtsdemux have a new pad, it calls the function on_new_pad: > in case of video it completes the pipeline with: > queue ! mpeg2dec ! videoscale ! capsfilter ! glupload ! fakesink (with > sync=true) > > in case of dsmcc content: > queue ! appsink > > The problem is that the video part plays faster than the usual, > nevertheless if I comment all the dsmcc part on "on_pad_added" function, > video part plays correctly. > > I can't found where is the problem, but I think it is related with the > dsmcc part pipeline, here you can found the link to "on_pad_added" function: > http://pastebin.ca/1550152 > > Thanks, > > Miquel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From tristan at sat.qc.ca Tue Sep 1 22:01:36 2009 From: tristan at sat.qc.ca (Tristan Matthews) Date: Tue, 01 Sep 2009 16:01:36 -0400 Subject: [gst-devel] gstrtpbin and multicast Message-ID: <4A9D7DA0.5020602@sat.qc.ca> Hi, My question is about using gstrtbin for multicast streaming. To adapt the examples from gst-plugins-good/test/examples/rtp/, such as server-v4l2-H264-alsasrc-PCMA.sh and client-H264-PCMA.sh, is it sufficient to specify the "multicast-group" and "multicast-interface" properties on the client's udpsrc elements and in the server use the multicast-group address as the udpsink elements' host property? Will rtp/rtcp still work properly for the case where i have one server and two or more clients? I've tested this out on a three machine setup and wireshark tells me that each client is sending receiver reports and that the server is sending sender reports, and in general it looks fine. I just wanted to make sure I wasn't missing any other critical steps. Thanks, Tristan -- Tristan Matthews Soci?t? des arts technologiques [SAT] email: tristan at sat.qc.ca web: http://www.music.mcgill.ca/~tmatthews From bilboed at gmail.com Wed Sep 2 18:26:37 2009 From: bilboed at gmail.com (Edward Hervey) Date: Wed, 02 Sep 2009 18:26:37 +0200 Subject: [gst-devel] GNonlin 0.10.13.3 pre-release Message-ID: <1251908797.26471.2.camel@localhost> Hi all, I've just pushed another GNonLin pre-release which fixes a long-lasting QoS issue. You'll find the 0.10.12.3 tarballs here: http://gstreamer.org/data/src/gnonlin/pre/ Hopefully this is the last pre-release before a release. I'm looking at doing it around Friday or Saturday. As usual, feedback, comments, and bug reports are more than welcome. Edward From sebastian.droege at collabora.co.uk Thu Sep 3 12:39:58 2009 From: sebastian.droege at collabora.co.uk (Sebastian =?ISO-8859-1?Q?Dr=F6ge?=) Date: Thu, 03 Sep 2009 12:39:58 +0200 Subject: [gst-devel] RELEASE: GStreamer C# bindings 0.9.0 "Good things come to those who wait" Message-ID: <1251974398.4886.48.camel@odin.lan> This mail announces the release of the GStreamer C# bindings 0.9.0 "Good things come to those who wait". The GStreamer C# Bindings are bindings for the GStreamer 0.10 release series and selected libraries and plugins. It comes with a number of examples. For more information, see http://gstreamer.freedesktop.org/modules/gstreamer-sharp.html To file bugs, go to http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gst-sharp Enjoy! PS: Note that this release doesn't make any API/ABI stability guarantees, see the release notes and the README for more information! -------------- next part -------------- Release notes for GStreamer C# bindings?0.9.0 "Good things come to those who wait" The GStreamer team is proud to announce a first development release of the GStreamer C# bindings for the 0.10.x release series. The 0.10.x series is a stable series targeted at end users. It is not API or ABI compatible with the stable 0.8.x series. It is, however, parallel installable with the 0.8.x series. Please note that at this time the GStreamer C# bindings are not consindered API/ABI stable, and public interfaces may still change from release to release. These changes will most likely be small. Please read the README file for more information on this. Features of this release * Initial release of gstreamer-sharp There were no bugs fixed in this release Download You can find source releases of gstreamer-sharp in the download directory: http://gstreamer.freedesktop.org/src/gstreamer-sharp/ GStreamer Homepage More details can be found on the project's website: http://gstreamer.freedesktop.org/ Support and Bugs We use GNOME's bugzilla for bug reports and feature requests: http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer Developers GStreamer is stored in Git, hosted at git.freedesktop.org, and can be cloned from there. Interested developers of the core library, plug-ins, and applications should subscribe to the gstreamer-devel list. If there is sufficient interest we will create more lists as necessary. Applications Contributors to this release * Aaron Bockover * Khaled Mohammed * Maarten Bosmans * Michael Dominic K * Sebastian Dr?ge ? -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: Dies ist ein digital signierter Nachrichtenteil URL: From fmogollon at vicomtech.org Fri Sep 4 10:58:22 2009 From: fmogollon at vicomtech.org (Felipe =?ISO-8859-1?Q?Mogoll=F3n?=) Date: Fri, 04 Sep 2009 10:58:22 +0200 Subject: [gst-devel] Mute stream while playing over RTSP server Message-ID: <1252054702.3262.37.camel@debian> Hello, I have build a RSTP server using gst-rtsp-server. I play a stream over RTSP using a pipeline like this: filesrc location=/home/usuario/myvideo.avi ! decodebin name=dec dec. ! glupload ! glfiltercube ! gldownload ! x264enc bitrate=100 ! rtph264pay name=pay0 pt=96 dec. ! queue ! audioresample ! audioconvert ! volume ! audioconvert ! alawenc ! rtppcmapay name=pay1 pt=90 I get volume plugin and I set mute parameter to TRUE in this way: gboolean oldVolume; gboolean mutedVolume=TRUE; gboolean newVolume; g_object_get(myVolume, "mute", &oldVolume, NULL); g_debug("Previous volume %d\n",oldVolume); g_object_set(myVolume, "mute", mutedVolume, NULL); g_object_get(myVolume, "mute", &newVolume, NULL); g_debug("Modified volume %d\n",newVolume); It prints that mute has been modified and set to 1, but when I listen the stream using vlc I can hear sound. Thanks in advance, Felipe From walter.kulecz-1 at nasa.gov Fri Sep 4 15:47:40 2009 From: walter.kulecz-1 at nasa.gov (Kulecz, Walter (JSC-SK)[WYLE LABORATORIES]) Date: Fri, 4 Sep 2009 08:47:40 -0500 Subject: [gst-devel] Viable strategy? Message-ID: <1F2262C21D3BB74DBE5C9576EAB4BF4F944D3AC7CD@NDJSSCC03.ndc.nasa.gov> I'd like some advice about how viable this pipeline layout would be and if I'm missing anything: pipeline1: vrl2src->tee->queue->myImageProcessingPlugin->xvimagesink pipeline2: \queue->mjpegencode->filesink Pipeline1 would run continuously from program startup till termination for real-time monitoring and analysis. Pipeline2 would be started and stopped as needed to record various epochs of interest. The main reason I ask is, if you add audio to it, you have a Linux video capture application, of which I've yet to find a good one (other than perhaps systems like MythTV which I've not actually used) so if gstreamer would make it this "easy" I have to ask why not one or more already?. Any issues I need to be aware of starting and stopping pipeline2? Glitch free operation is pretty important. The reason for pipeline2 is because I would eventually write an "off-line" application for attempting to handle segments where the real-time tracking failed due to transient lighting changes, etc. that perhaps could be fixed with changes to the analysis parameters. Pipeline3: filesrc->myImageProcessingPlugin ->xvimagesink. What we do now is split the video and run it into my version1 image processing application and record to standalone S-VHS or DVD recorders in parallel. For the off-line step my version1 app uses the output of the recorder instead of live video. This does ok, but the workflow is tedious. Any advice as to ximagesink vs. xvimagesink? I'd like to display the video output within a gtk window, I've found a sample code that sort of works and uses ximagesink. The long term goal would be to add analog data to the stream captured along with the video. One obvious possibility would be to abuse the audio subsystems of gstreamer since things like alsasrc with audio/x-raw-int claim to support sample rates and number of channels from 1-2^31, I'd be happy with a max 16-20 channels at 2000Hz which would only be about 80K bytes/sec aggregate rate for 16-bit data which is a bit lower than "normal" CD quality stereo. From t.i.m at zen.co.uk Fri Sep 4 16:56:52 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Fri, 04 Sep 2009 15:56:52 +0100 Subject: [gst-devel] play MOV file In-Reply-To: <20090829172123.2e4982fc.dick@nagual.nl> References: <20090829172123.2e4982fc.dick@nagual.nl> Message-ID: <1252076212.4753.9.camel@zingle> On Sat, 2009-08-29 at 17:21 +0200, dick hoogendijk wrote: > I used to be able to play this .mov file with the totem plugin and > firefox. This was on OpenSolaris-B118. After upgrading to B121 I get an > error message saying: > > The playback of this movie requires a text/html decoder plugin which is > not installed. > > Has this file gone missing in the upgrade process? > Any idea which file this is (I may recover it from B118) Sounds like you might be running into: http://bugzilla.gnome.org/show_bug.cgi?id=592665 http://bugzilla.gnome.org/show_bug.cgi?id=375867 Cheers -Tim From francis.meyvis at gmail.com Fri Sep 4 17:23:08 2009 From: francis.meyvis at gmail.com (franchan) Date: Fri, 4 Sep 2009 17:23:08 +0200 Subject: [gst-devel] audio playback shift in time Message-ID: <8456544a0909040823g6e65869fx1201bab2a11421bc@mail.gmail.com> Hello, I have a custom mpegts demux. It gets data through the appsrc. Into the appsrc I push the mpegts coming from an UDP socket. This allows to do changing from UDP socket (other mpegts stream). When this happens I change the playbin2 first from playing to ready, Then back to playing again. The mpegts demux, when it sees the state change, removes its pads and waits for the new PAT/PMT information. When the mpegts sees the new PAT PMT, it recreates the new pads. Before the mpegts plugin sends out the PES data, I first send a flush-start, flush-stop and the new_segment event downstream. with the start and position time corresponding to the first PTS detected in the new mpegts data. For the first socket, when starting playback, all works fine: audio and video are in sync. When switching the socket the video starts playing fine after 1 to 2 seconds. However the audio behaves odd: its playback is delayed by the amount of time socket 1 has been playing. I verified the PTS timestamps on the input (in mpegts demux) and the output (audio/video decoders). These all seem fine; audio/video data is alternately push forward into the sinks. Still the alsa sink (or something else after the audio decoder) introduces this strange delay. Can somebody tell me what is going on and how to remove this delay for audio and still having the video play back as well? Thanks, francis From jcastellanos77 at gmail.com Fri Sep 4 19:31:32 2009 From: jcastellanos77 at gmail.com (Joaquin Castellanos) Date: Fri, 4 Sep 2009 12:31:32 -0500 Subject: [gst-devel] Info for codec_data parsing for H264 in a FLV container Message-ID: <400edb7e0909041031q40f71641t8b4b738b85d465ec@mail.gmail.com> Hi I am looking for information required to parse H264 codec_data. With some Flv containers (with H264 v-streams) the flvdemux does not parse width, height or framerate, instead the demuxer sent the codec_data to the next element. e.g. # gst-launch filesrc location =/data/EVM_filesystems/x0089714/target/Vid.flv ! flvdemux name=demux demux.video ! fakesink -v Setting pipeline to PAUSED ... /GstPipeline:pipeline0/GstFLVDemux:demux.GstPad:video: caps = video/x-h264, pixel-aspect-ratio=(fraction)1/1, codec_data=(buffer)01640033ffe1001c67640033ac2cc502d0ceffc01400144400000fa40003a9823c60c65801000468eebcb0 Best regards Joaquin -------------- next part -------------- An HTML attachment was scrubbed... URL: From jam at smru.co.uk Sat Sep 5 22:22:23 2009 From: jam at smru.co.uk (00a) Date: Sat, 5 Sep 2009 13:22:23 -0700 (PDT) Subject: [gst-devel] Find out what sinks are avaiable. Message-ID: <25254429.post@talk.nabble.com> If I have several usb sound cards how do I select which one to output to? I have installed gstreamer on my embedded system and can hear the tone from by default headphone socket using gst-launch-0.10 audiotestsrc ! audioconvert ! audioresample ! alsasink I tried adding device=1 at the end but this did nothing apart from error. Can someone help me out with the syntax and the ability to query what] soundcards are available? -- View this message in context: http://www.nabble.com/Find-out-what-sinks-are-avaiable.-tp25254429p25254429.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From yiliangb at gmail.com Sat Sep 5 23:06:48 2009 From: yiliangb at gmail.com (Yiliang Bao) Date: Sat, 5 Sep 2009 14:06:48 -0700 (PDT) Subject: [gst-devel] how to cleanly re-install gstreamer? Help!! Message-ID: <25252421.post@talk.nabble.com> Hi, I compiled and installed the latest gst-rtsp-server, and updated gstreamer itself based on the error messages from gst-rtsp-server installation. However, after these steps, I could no longer find some plugins like mad, xvimagesink, v4l2src, etc. I have tried re-installing the latest versions of all plugin packages. I have also tried to remove, then re-install all plugins from Synaptic Package Manager. Got some plugin, like v4l2src, back, but mad, xvimagesink are still missing. autovideosink is there, but some pipleine which works is no longer working. For example: gst-launch v4l2src device=/dev/video0 ! autovideosink >>>>>> Error messages: Setting pipeline to PAUSED ... ERROR: Pipeline doesn't want to pause. ERROR: from element /GstPipeline:pipeline0/GstAutoVideoSink:autovideosink0: Could not initialize supporting library. Additional debug info: gstautovideosink.c(373): gst_auto_video_sink_detect (): /GstPipeline:pipeline0/GstAutoVideoSink:autovideosink0: Failed to set target pad Setting pipeline to NULL ... Freeing pipeline ... gst-launch filesrc location=Videos/xyz.avi ! decodebin ! autovideosink >>>>>>>>> Error messages: Setting pipeline to PAUSED ... Pipeline is PREROLLING ... ERROR: from element /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstAviDemux:avidemux0: Internal data stream error. Additional debug info: gstavidemux.c(4443): gst_avi_demux_loop (): /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstAviDemux:avidemux0: streaming stopped, reason not-linked ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... Freeing pipeline ... Does anyone know why? Is it possible to re-install a clean version of gstreamer? Yiliang -- View this message in context: http://www.nabble.com/how-to-cleanly-re-install-gstreamer--Help%21%21-tp25252421p25252421.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From vitaly.v.ch at gmail.com Sun Sep 6 13:13:42 2009 From: vitaly.v.ch at gmail.com (Vitaly V. Ch) Date: Sun, 6 Sep 2009 14:13:42 +0300 Subject: [gst-devel] ERROR: from element /pipeline0/timidity0: Could not decode stream. Message-ID: <6efe08af0909060413u42207b0evff87f2a512e010f1@mail.gmail.com> I want to play midi via gstreamer but got next trouble: # LANG= gst-launch filesrc location=Krysha_doma_tvoego.mid ! timidity ! alsasink Setting pipeline to PAUSED ... Pipeline is PREROLLING ... ERROR: from element /pipeline0/timidity0: Could not decode stream. Additional debug info: gsttimidity.c(641): gst_timidity_loop (): /pipeline0/timidity0: Unable to parse midi ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... FREEING pipeline ... # From ensonic at hora-obscura.de Sun Sep 6 13:39:30 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sun, 06 Sep 2009 14:39:30 +0300 Subject: [gst-devel] ANN: buzztard 0.5.0 "crown of thorns" Message-ID: <4AA39F72.6070306@hora-obscura.de> hi, The buzztard team has released version 0.5.0 "crown of thorns" of its buzz-alike music composer. All modules got extensive improvements over the last release from almost a year ago. Give it a try, join hacking and report bugs. bml -------------------------------------------------------------------------------- Split the api into library and instance api. Lots of code and build cleanups. bsl -------------------------------------------------------------------------------- Several bug fixes and better compatibility. buzztard -------------------------------------------------------------------------------- Rewrite of internal pipeline management. One can now play partially connected songs, and add/remove plugins while playing. Buzztard can play notes while editing. More robust saving of songs. Lots of bugfixes and UI improvements. Better user-guide including three small tutorials. Initial support for python and javascript via gobject introspection. gst-buzztard -------------------------------------------------------------------------------- Lots of improvements in the buzzmachine wrapper. Buzz index support for categories. Make machines live-playable. More resource efficient. project-page: http://www.buzztard.org screenshots: http://www.buzztard.org/index.php/Screenshots downloads : http://sourceforge.net/project/showfiles.php?group_id=55124 buzztard core developer team -- http://www.buzztard.org From ensonic at hora-obscura.de Sun Sep 6 21:15:31 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sun, 06 Sep 2009 22:15:31 +0300 Subject: [gst-devel] ERROR: from element /pipeline0/timidity0: Could not decode stream. In-Reply-To: <6efe08af0909060413u42207b0evff87f2a512e010f1@mail.gmail.com> References: <6efe08af0909060413u42207b0evff87f2a512e010f1@mail.gmail.com> Message-ID: <4AA40A53.4050409@hora-obscura.de> Vitaly V. Ch schrieb: > I want to play midi via gstreamer but got next trouble: > > # LANG= gst-launch filesrc location=Krysha_doma_tvoego.mid ! timidity ! alsasink > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > ERROR: from element /pipeline0/timidity0: Could not decode stream. > Additional debug info: > gsttimidity.c(641): gst_timidity_loop (): /pipeline0/timidity0: > Unable to parse midi > ERROR: pipeline doesn't want to preroll. > Setting pipeline to NULL ... > FREEING pipeline ... > # > can you play that with other players. Please double check that its a midi-file (what does "file Krysha_doma_tvoego.mid" reports). If it only fails with gstreamer, please file a bug and attach the file. Stefan From vitaly.v.ch at gmail.com Mon Sep 7 08:50:01 2009 From: vitaly.v.ch at gmail.com (Vitaly V. Ch) Date: Mon, 7 Sep 2009 09:50:01 +0300 Subject: [gst-devel] ERROR: from element /pipeline0/timidity0: Could not decode stream. In-Reply-To: <4AA40A53.4050409@hora-obscura.de> References: <6efe08af0909060413u42207b0evff87f2a512e010f1@mail.gmail.com> <4AA40A53.4050409@hora-obscura.de> Message-ID: <6efe08af0909062350w31705eb7n1f1e58d4a27f6169@mail.gmail.com> This midi file played successfully via aplaymidi but on my system gstreamer can't play midi files at all. I'm beginner in Gstreamer and in this case need guide. \\wbr Vitaly On Sun, Sep 6, 2009 at 10:15 PM, Stefan Kost wrote: > Vitaly V. Ch schrieb: >> I want to play midi via gstreamer but got next trouble: >> >> # LANG= gst-launch filesrc location=Krysha_doma_tvoego.mid ! timidity ! alsasink >> Setting pipeline to PAUSED ... >> Pipeline is PREROLLING ... >> ERROR: from element /pipeline0/timidity0: Could not decode stream. >> Additional debug info: >> gsttimidity.c(641): gst_timidity_loop (): /pipeline0/timidity0: >> Unable to parse midi >> ERROR: pipeline doesn't want to preroll. >> Setting pipeline to NULL ... >> FREEING pipeline ... >> # >> > > can you play that with other players. Please double check that its a midi-file > (what does "file Krysha_doma_tvoego.mid" reports). If it only fails with > gstreamer, please file a bug and attach the file. > > Stefan > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. ?http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From t.i.m at zen.co.uk Mon Sep 7 09:38:42 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Mon, 07 Sep 2009 08:38:42 +0100 Subject: [gst-devel] Viable strategy? In-Reply-To: <1F2262C21D3BB74DBE5C9576EAB4BF4F944D3AC7CD@NDJSSCC03.ndc.nasa.gov> References: <1F2262C21D3BB74DBE5C9576EAB4BF4F944D3AC7CD@NDJSSCC03.ndc.nasa.gov> Message-ID: <1252309122.5230.14.camel@zingle> On Fri, 2009-09-04 at 08:47 -0500, Kulecz, Walter wrote: > I'd like some advice about how viable this pipeline layout would be and if I'm missing anything: > > pipeline1: vrl2src->tee->queue->myImageProcessingPlugin->xvimagesink > pipeline2: \queue->mjpegencode->filesink > > Pipeline1 would run continuously from program startup till termination > for real-time monitoring and analysis. Pipeline2 would be started and > stopped as needed to record various epochs of interest. Looks ok. The main problem with such setups is usually that if you link in the second part mid-stream you won't get a newsegment event and timestamps starting at a non-zero value, which often causes problem when saving to a container format. If you're just dumping video frames as MJPEG, that shouldn't be an issue though. You might also want to have a look at camerabin from gst-plugins-bad. > Any advice as to ximagesink vs. xvimagesink? I'd like to display the > video output within a gtk window, I've found a sample code that sort > of works and uses ximagesink. Both should behave pretty much the same with regard to GstXOverlay. ximagesink usually requires RGB input and doesn't do scaling, whereas xvimagesink accepts YUV and does scale. The downside with xvimagesink is that you can usually only have one at a time (depending on hardware/drivers), or can't even use it at all on some systems (if the driver doesn't support it). ximagesink should always work as long as you're using X11. Cheers -Tim From Jerry.Tan at Sun.COM Mon Sep 7 09:53:06 2009 From: Jerry.Tan at Sun.COM (Jerry Tan) Date: Mon, 07 Sep 2009 15:53:06 +0800 Subject: [gst-devel] Does gstreamer support audio with 4-bit G.721 ADPCM? Message-ID: <4AA4BBE2.20102@sun.com> it seems that it does not. From t.i.m at zen.co.uk Mon Sep 7 10:09:13 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Mon, 07 Sep 2009 09:09:13 +0100 Subject: [gst-devel] Does gstreamer support audio with 4-bit G.721 ADPCM? In-Reply-To: <4AA4BBE2.20102@sun.com> References: <4AA4BBE2.20102@sun.com> Message-ID: <1252310953.5230.15.camel@zingle> On Mon, 2009-09-07 at 15:53 +0800, Jerry Tan wrote: > it seems that it does not. If you have a file that doesn't play, please file a bug in bugzilla, thanks! Cheers -Tim From bilboed at gmail.com Mon Sep 7 12:07:53 2009 From: bilboed at gmail.com (Edward Hervey) Date: Mon, 07 Sep 2009 12:07:53 +0200 Subject: [gst-devel] RELEASE: GStreamer GNonLin plugins 0.10.13 "Service of Quality" Message-ID: <1252318073.16797.3.camel@localhost> This mail announces the release of GNonLin GStreamer plugins 0.10.13 "Service of Quality" This module contains a set of plug-ins for GStreamer to ease the creation of multimedia editors, or any other application where a timeline-oriented use of GStreamer makes sense. For more information, see http://gstreamer.freedesktop.org/modules/gnonlin.html To file bugs, go to http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gnonlin -------------- next part -------------- Release notes for GNonLin Non-Linear Editing Plug-ins?0.10.13 "Service of Quality" The GStreamer team is proud to announce a new release in the 0.10.x stable series of GNonLin. This module contains a set of plug-ins for GStreamer to ease the creation of multimedia editors, or any other application where a timeline-oriented use of GStreamer makes sense. These elements include: gnlsource An element for using source elements/bins in a GnlComposition gnlfilesource A higher-level element for using a uri in a GnlComposition gnlcomposition A container element that handles GNonLin objects gnloperation An element for using filters in a GnlComposition Features of this release * Fix QoS event handling * Fix racyness in source pad handlings * GnlOperation: Add signal to know input stream priorities Bugs fixed in this release * 583145 : Seeking on pending pipelines should return True. Download You can find source releases of gnonlin in the download directory: http://gstreamer.freedesktop.org/src/gnonlin/ GStreamer Homepage More details can be found on the project's website: http://gstreamer.freedesktop.org/ Support and Bugs We use GNOME's bugzilla for bug reports and feature requests: http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer Developers GStreamer is stored in Git, hosted at git.freedesktop.org, and can be cloned from there. Interested developers of the core library, plug-ins, and applications should subscribe to the gstreamer-devel list. If there is sufficient interest we will create more lists as necessary. Applications GNonLin is primarily used by PiTiVi (http://www.pitivi.org/) and Jokosher (http://www.jokosher.org/). Contributors to this release * Adam Dingle * Alessandro Decina * Edward Hervey * Jan Schmidt * Olivier Cr?te * Stefan Kost ? From Jerry.Tan at Sun.COM Tue Sep 8 05:05:00 2009 From: Jerry.Tan at Sun.COM (Jerry Tan) Date: Tue, 08 Sep 2009 11:05:00 +0800 Subject: [gst-devel] Does gstreamer support audio with 4-bit G.721 ADPCM? In-Reply-To: <1252310953.5230.15.camel@zingle> References: <4AA4BBE2.20102@sun.com> <1252310953.5230.15.camel@zingle> Message-ID: <4AA5C9DC.3060300@sun.com> I file a bug there http://bugzilla.gnome.org/show_bug.cgi?id=594454 and attached the audio file there also. Tim-Philipp M?ller : > On Mon, 2009-09-07 at 15:53 +0800, Jerry Tan wrote: > > >> it seems that it does not. >> > > If you have a file that doesn't play, please file a bug in bugzilla, > thanks! > > Cheers > -Tim > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From rakesh2.sharma at aricent.com Tue Sep 8 08:47:58 2009 From: rakesh2.sharma at aricent.com (rakesh sharma) Date: Mon, 7 Sep 2009 23:47:58 -0700 (PDT) Subject: [gst-devel] CPU Load bcoz of Gstreamer plugin Message-ID: <25340716.post@talk.nabble.com> I have written mp4sp encoder and its tacking 99% cpu load for all resolutions. Can anyone suggest solution for this ? Is that i have to release cpu using some gstreamer api ? Please help me its urgent..... -- View this message in context: http://www.nabble.com/CPU-Load-bcoz-of-Gstreamer-plugin-tp25340716p25340716.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From bilboed at gmail.com Tue Sep 8 10:13:28 2009 From: bilboed at gmail.com (Edward Hervey) Date: Tue, 08 Sep 2009 10:13:28 +0200 Subject: [gst-devel] CPU Load bcoz of Gstreamer plugin In-Reply-To: <25340716.post@talk.nabble.com> References: <25340716.post@talk.nabble.com> Message-ID: <1252397608.2148.7.camel@putamadre> Sorry to ask the obvious question... ... but have you profiled your pipeline and/or encoder to see *WHERE* it's taking 99% cpu ? Use a profiler like callgrind/oprofile (if available on your platform) or analyze your logs to see where it's taking all that time. Until you've analyzed where your problem is... we won't be able to help you. On Mon, 2009-09-07 at 23:47 -0700, rakesh sharma wrote: > I have written mp4sp encoder and its tacking 99% cpu load for all > resolutions. > Can anyone suggest solution for this ? > Is that i have to release cpu using some gstreamer api ? > Please help me its urgent..... From mail at renestadler.de Tue Sep 8 10:54:02 2009 From: mail at renestadler.de (=?ISO-8859-1?Q?Ren=E9_Stadler?=) Date: Tue, 08 Sep 2009 11:54:02 +0300 Subject: [gst-devel] CPU Load bcoz of Gstreamer plugin In-Reply-To: <25340716.post@talk.nabble.com> References: <25340716.post@talk.nabble.com> Message-ID: <4AA61BAA.2050503@renestadler.de> rakesh sharma schrieb: > I have written mp4sp encoder and its tacking 99% cpu load for all > resolutions. > Can anyone suggest solution for this ? > Is that i have to release cpu using some gstreamer api ? > Please help me its urgent..... Obvious question, but: Is it that you run this with a non-live pipeline? E.g. videotestsrc ! mp4spenc ! ... or transcoding. In this case it is natural for the pipeline to take up 100% of CPU since it runs as fast as possible... You need to limit the rate at the source by using a live source (like videotestsrc is-live=true) or have the sink perform time sync (encoder ! fakesink sync=true will do, assuming you forward the input timestamps correctly). -- Regards, Ren? Stadler From rakesh2.sharma at aricent.com Tue Sep 8 12:44:11 2009 From: rakesh2.sharma at aricent.com (rakesh sharma) Date: Tue, 8 Sep 2009 03:44:11 -0700 (PDT) Subject: [gst-devel] CPU Load bcoz of Gstreamer plugin In-Reply-To: <1252397608.2148.7.camel@putamadre> References: <25340716.post@talk.nabble.com> <1252397608.2148.7.camel@putamadre> Message-ID: <25343663.post@talk.nabble.com> We dont have those tools callgrind/oprofile on the target platform. I have written the plugin in lcml(socket node), so code should take maximum cpu on dsp side not arm side, but its tacking full cpu on arm side . I am not able to measure where its eating cpu. If you people have those tools for arm you can share it with me so that i'll check and let you know about that.please reply soon its urgent. thanks for reply... Edward Hervey wrote: > > Sorry to ask the obvious question... > > ... but have you profiled your pipeline and/or encoder to see *WHERE* > it's taking 99% cpu ? Use a profiler like callgrind/oprofile (if > available on your platform) or analyze your logs to see where it's > taking all that time. > > Until you've analyzed where your problem is... we won't be able to > help you. > > On Mon, 2009-09-07 at 23:47 -0700, rakesh sharma wrote: >> I have written mp4sp encoder and its tacking 99% cpu load for all >> resolutions. >> Can anyone suggest solution for this ? >> Is that i have to release cpu using some gstreamer api ? >> Please help me its urgent..... > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 > 30-Day > trial. Simplify your report design, integration and deployment - and focus > on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- View this message in context: http://www.nabble.com/CPU-Load-bcoz-of-Gstreamer-plugin-tp25340716p25343663.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From bilboed at gmail.com Tue Sep 8 13:07:38 2009 From: bilboed at gmail.com (Edward Hervey) Date: Tue, 08 Sep 2009 13:07:38 +0200 Subject: [gst-devel] CPU Load bcoz of Gstreamer plugin In-Reply-To: <25343663.post@talk.nabble.com> References: <25340716.post@talk.nabble.com> <1252397608.2148.7.camel@putamadre> <25343663.post@talk.nabble.com> Message-ID: <1252408058.16797.24.camel@localhost> On Tue, 2009-09-08 at 03:44 -0700, rakesh sharma wrote: > We dont have those tools callgrind/oprofile on the target platform. > I have written the plugin in lcml(socket node), so code should take maximum > cpu on dsp side not arm side, but its tacking full cpu on arm side . I am > not able to measure where its eating cpu. > If you people have those tools for arm you can share it with me so that i'll > check and let you know about that.please reply soon its urgent. 1. I just can not believe your employer doesn't have profiling tools for ARM. (Googling for "arm profiling linux" returns heaps of results/tools). 2. If you'd carried on reading my mail you would have read "or analyze your logs to see where it's taking all that time". By logs, I mean GST_DEBUG logs. Edward > > thanks for reply... > > > Edward Hervey wrote: > > > > Sorry to ask the obvious question... > > > > ... but have you profiled your pipeline and/or encoder to see *WHERE* > > it's taking 99% cpu ? Use a profiler like callgrind/oprofile (if > > available on your platform) or analyze your logs to see where it's > > taking all that time. > > > > Until you've analyzed where your problem is... we won't be able to > > help you. > > > > On Mon, 2009-09-07 at 23:47 -0700, rakesh sharma wrote: > >> I have written mp4sp encoder and its tacking 99% cpu load for all > >> resolutions. > >> Can anyone suggest solution for this ? > >> Is that i have to release cpu using some gstreamer api ? > >> Please help me its urgent..... > > > > > > ------------------------------------------------------------------------------ > > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 > > 30-Day > > trial. Simplify your report design, integration and deployment - and focus > > on > > what you do best, core application coding. Discover what's new with > > Crystal Reports now. http://p.sf.net/sfu/bobj-july > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > From marlene.hildebrandehrhardt at gmail.com Tue Sep 8 15:14:04 2009 From: marlene.hildebrandehrhardt at gmail.com (=?ISO-8859-1?Q?Marl=E8ne_Hildebrand=2DEhrhardt?=) Date: Tue, 8 Sep 2009 15:14:04 +0200 Subject: [gst-devel] What does this GStreamer error mean? Message-ID: <92f1bb260909080614t433d3208wbac0162f6637ee1d@mail.gmail.com> Hi! I'm using a streaming application that has been developped with Java and GStreamer. Sometimes, when I launch the application, after a few streams have been created, the application crashes and I get the following message : [java] [java] (JavaUBIK:23929): GStreamer-CRITICAL **: [java] Trying to dispose element udpsrc34, but it is not in the NULL state. [java] You need to explicitly set elements to the NULL state before [java] dropping the final reference, to allow them to clean up. [java] [java] [java] GThread-ERROR **: file /build/buildd/glib2.0-2.20.1/gthread/gthread-posix.c: line 171 (g_mutex_free_posix_impl): error 'P?riph?rique ou ressource occup?' during 'pthread_mutex_destroy ((pthread_mutex_t *) mutex)' [java] aborting... [java] Java Result: 134 Does anybody have an idea of the signification of this message? Thank you by advance! Marl?ne Hildebrand-Ehrhardt -------------- next part -------------- An HTML attachment was scrubbed... URL: From rakesh2.sharma at aricent.com Tue Sep 8 15:15:28 2009 From: rakesh2.sharma at aricent.com (rakesh sharma) Date: Tue, 8 Sep 2009 06:15:28 -0700 (PDT) Subject: [gst-devel] CPU Load bcoz of Gstreamer plugin In-Reply-To: <1252408058.16797.24.camel@localhost> References: <25340716.post@talk.nabble.com> <1252397608.2148.7.camel@putamadre> <25343663.post@talk.nabble.com> <1252408058.16797.24.camel@localhost> Message-ID: <25346008.post@talk.nabble.com> On device there is no debug mode supported. Thanks for that googling thing i'll search for those tools. Any other idea... Edward Hervey wrote: > > On Tue, 2009-09-08 at 03:44 -0700, rakesh sharma wrote: >> We dont have those tools callgrind/oprofile on the target platform. >> I have written the plugin in lcml(socket node), so code should take >> maximum >> cpu on dsp side not arm side, but its tacking full cpu on arm side . I am >> not able to measure where its eating cpu. >> If you people have those tools for arm you can share it with me so that >> i'll >> check and let you know about that.please reply soon its urgent. > > 1. I just can not believe your employer doesn't have profiling tools for > ARM. (Googling for "arm profiling linux" returns heaps of > results/tools). > > 2. If you'd carried on reading my mail you would have read "or analyze > your logs to see where it's taking all that time". By logs, I mean > GST_DEBUG logs. > > Edward > >> >> thanks for reply... >> >> >> Edward Hervey wrote: >> > >> > Sorry to ask the obvious question... >> > >> > ... but have you profiled your pipeline and/or encoder to see *WHERE* >> > it's taking 99% cpu ? Use a profiler like callgrind/oprofile (if >> > available on your platform) or analyze your logs to see where it's >> > taking all that time. >> > >> > Until you've analyzed where your problem is... we won't be able to >> > help you. >> > >> > On Mon, 2009-09-07 at 23:47 -0700, rakesh sharma wrote: >> >> I have written mp4sp encoder and its tacking 99% cpu load for all >> >> resolutions. >> >> Can anyone suggest solution for this ? >> >> Is that i have to release cpu using some gstreamer api ? >> >> Please help me its urgent..... >> > >> > >> > >> ------------------------------------------------------------------------------ >> > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 >> > 30-Day >> > trial. Simplify your report design, integration and deployment - and >> focus >> > on >> > what you do best, core application coding. Discover what's new with >> > Crystal Reports now. http://p.sf.net/sfu/bobj-july >> > _______________________________________________ >> > gstreamer-devel mailing list >> > gstreamer-devel at lists.sourceforge.net >> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > >> > >> > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 > 30-Day > trial. Simplify your report design, integration and deployment - and focus > on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- View this message in context: http://www.nabble.com/CPU-Load-bcoz-of-Gstreamer-plugin-tp25340716p25346008.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From yiliangb at gmail.com Wed Sep 2 08:41:30 2009 From: yiliangb at gmail.com (Yiliang Bao) Date: Tue, 1 Sep 2009 23:41:30 -0700 (PDT) Subject: [gst-devel] how to cleanly re-install gstreamer? Help!! Message-ID: <25252421.post@talk.nabble.com> Hi, I compiled and installed the latest gst-rtsp-server, and updated gstreamer itself based on the error messages from gst-rtsp-server installation. However, after these steps, I could no longer find some plugins like mad, xvimagesink, v4l2src, etc. I have tried re-installing the latest versions of all plugin packages. I have also tried to remove, then re-install all plugins from Synaptic Package Manager. Got some plugin, like v4l2src, back, but mad, xvimagesink are still missing. autovideosink is there, but some pipleine which works is no longer working. For example: gst-launch v4l2src device=/dev/video0 ! autovideosink >>>>>> Error messages: Setting pipeline to PAUSED ... ERROR: Pipeline doesn't want to pause. ERROR: from element /GstPipeline:pipeline0/GstAutoVideoSink:autovideosink0: Could not initialize supporting library. Additional debug info: gstautovideosink.c(373): gst_auto_video_sink_detect (): /GstPipeline:pipeline0/GstAutoVideoSink:autovideosink0: Failed to set target pad Setting pipeline to NULL ... Freeing pipeline ... gst-launch filesrc location=Videos/xyz.avi ! decodebin ! autovideosink >>>>>>>>> Error messages: Setting pipeline to PAUSED ... Pipeline is PREROLLING ... ERROR: from element /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstAviDemux:avidemux0: Internal data stream error. Additional debug info: gstavidemux.c(4443): gst_avi_demux_loop (): /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstAviDemux:avidemux0: streaming stopped, reason not-linked ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... Freeing pipeline ... Does anyone know why? Is it possible to re-install a clean version of gstreamer? Yiliang -- View this message in context: http://www.nabble.com/how-to-cleanly-re-install-gstreamer--Help%21%21-tp25252421p25252421.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From jam at smru.co.uk Wed Sep 2 11:34:56 2009 From: jam at smru.co.uk (00a) Date: Wed, 2 Sep 2009 02:34:56 -0700 (PDT) Subject: [gst-devel] Find out what sinks are avaiable. Message-ID: <25254429.post@talk.nabble.com> If I have several usb sound cards how do I select which one to output to? I have installed gstreamer on my embedded system and can hear the tone from by default headphone socket using gst-launch-0.10 audiotestsrc ! audioconvert ! audioresample ! alsasink I tried adding device=1 at the end but this did nothing apart from error. Can someone help me out with the syntax and the ability to query what] soundcards are available? -- View this message in context: http://www.nabble.com/Find-out-what-sinks-are-avaiable.-tp25254429p25254429.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From jam at smru.co.uk Fri Sep 4 00:20:24 2009 From: jam at smru.co.uk (00a) Date: Thu, 3 Sep 2009 15:20:24 -0700 (PDT) Subject: [gst-devel] Find out what sinks are avaiable. Message-ID: <25254429.post@talk.nabble.com> If I have several usb sound cards how do I select which one to output to? I have installed gstreamer on my embedded system and can hear the tone from by default headphone socket using gst-launch-0.10 audiotestsrc ! audioconvert ! audioresample ! alsasink I tried adding device=1 at the end but this did nothing apart from error. Can someone help me out with the syntax and the ability to query what] soundcards are available? -- View this message in context: http://www.nabble.com/Find-out-what-sinks-are-avaiable.-tp25254429p25254429.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From er.mayankkapoor at gmail.com Fri Sep 4 06:54:51 2009 From: er.mayankkapoor at gmail.com (co.sam) Date: Thu, 3 Sep 2009 21:54:51 -0700 (PDT) Subject: [gst-devel] new to gstreamer Message-ID: <25288295.post@talk.nabble.com> Hi friends, Hope you are doing great.Recently i started reading gstremer to play around with media files but I am absolutely new to gstreamer. I refered an application manual of gsteamer.Also i tried some gst-launch examples such as: 1.gst-launch audiotestsrc ! audioconvert ! audioresample ! osssink(plays sound) 2.gst-launch-0.10 ximagesrc num-buffers=1 ! ffmpegcolorspace ! pngenc ! filesink location=screenshot.png(captures the screen shot) Though I have tried these examples I have not understood the exact meaning of what these individual elements of this pipeline are donig and also how come these pipelines are constructed in this particular order. Please help in out to learn gstremer Any kind of help is welcome....... regards, co.sam -- View this message in context: http://www.nabble.com/new-to-gstreamer-tp25288295p25288295.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From gerecke at gmail.com Sun Sep 6 05:50:41 2009 From: gerecke at gmail.com (attaboy) Date: Sat, 5 Sep 2009 20:50:41 -0700 (PDT) Subject: [gst-devel] udpsrc multicast In-Reply-To: <3afe75670907211035kb4871abl76f7bb4a3d67ca57@mail.gmail.com> References: <3afe75670907211035kb4871abl76f7bb4a3d67ca57@mail.gmail.com> Message-ID: <25314433.post@talk.nabble.com> udpsrc is broken on Windows. There is a patch but I don't think it has made it into the trunk yet. http://bugzilla.gnome.org/show_bug.cgi?id=534243 Levi Pope wrote: > > I am trying to use udpsrc on windows to pull in a stream on a multicast > address > but all I get is this error. > > c:\gst-launch-0.10 --gst-debug=udpsrc:4 udpsrc multicast-group=224.1.2.3 > port=1234 ! fakesink dump=1 > > 0:00:00.093750000 4968 00352940 DEBUG udpsrc > gstudpsrc.c:611:gst_udpsrc_update_uri: updated uri to udp:// > 224.1.2.3 > :4951 > 0:00:00.093750000 4968 00352940 DEBUG udpsrc > gstudpsrc.c:611:gst_udpsrc_update_uri: updated uri to udp:// > 224.1.2.3 > :1234 > Setting pipeline to PAUSED ... > 0:00:00.125000000 4968 00352940 DEBUG udpsrc > gstudpsrc.c:806:gst_udpsrc_start: allocating socket for > 224.1.2.3:123 > 4 > 0:00:00.125000000 4968 00352940 DEBUG udpsrc > gstudpsrc.c:825:gst_udpsrc_start: binding on port 1234 > 0:00:00.125000000 4968 00352940 WARN udpsrc > gstudpsrc.c:967:gst_udpsrc_start: error: bind failed -1: No error > (0) > ERROR: Pipeline doesn't want to pause. > ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Could not > get/set settings from/on resource. > Additional debug info: > ..\..\..\Source\gst-plugins-good\gst\udp\gstudpsrc.c(967): > gst_udpsrc_start > (): /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: > bind failed -1: No error (0) > Setting pipeline to NULL ... > > Is there something I am missing? > > Thanks for the help > Levi > > ------------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- View this message in context: http://www.nabble.com/udpsrc-multicast-tp24592318p25314433.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From msauer2000 at yahoo.ca Mon Sep 7 22:16:01 2009 From: msauer2000 at yahoo.ca (Mark Sauer) Date: Mon, 7 Sep 2009 13:16:01 -0700 (PDT) Subject: [gst-devel] Problems with AV Sync with Open GL sink in Windows Message-ID: <25335761.post@talk.nabble.com> I have built gstreamer, and the plugins using mingw. I would like to use gstreamer to create a platform independent player. I have found video sinks that work great in linux. But in windows it seems that all the sinks I have tried have some problems. directdraw sink only takes rgb input, and does not support PAR corrections. dshowvideosink - I cannot build this no matter how hard I try with mingw due to the microsoft headers being nonstandard. I have built it with visual c++, but the resulting dll crashes gstreamer, so this is no good. sdlvideosink - I managed to build it, but it does not create a video surface in my test. glimagesink - I built this, and it seemed to be quite promising. It takes YUV input, supports PAR locking, and PAR settings from the video decoders. The problem I am having with glimagesink is that the audio and video tracks are not in sync. I am wondering if anyone has managed to get glimagesink to work on Windows with the AV in sync, and if so how this was done. And if not if anyone has advice on how to fix it, or what video output sync I should use for windows? It seems to me that a native Direct2D or Direct3D sync should be written for windows.. I am fine with using the glimagesink as well though, and seems to have some great features.. But obviously I need the video in sync with the audio. Depending on the content I have had different problems. One problem has the video play for about 2 seconds, and then it goes black, which seems like it is underrunning or buffering. The other problem has the video play fine along with the audio, but they are just out of sync (by about 1/2 second). I could correct the sync in this case using "render-delay" property. But I don't think sync should be a problem, as there was no av sync problem with the DirectDraw sink.. Thanks for any help. Mark Sauer -- View this message in context: http://www.nabble.com/Problems-with-AV-Sync-with-Open-GL-sink-in-Windows-tp25335761p25335761.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From dpalffy at rainstorm.org Tue Sep 8 16:28:17 2009 From: dpalffy at rainstorm.org (PALFFY Daniel) Date: Tue, 8 Sep 2009 16:28:17 +0200 (CEST) Subject: [gst-devel] timestamps on a live h264 source Message-ID: Hi, I'm developing a gstreamer source for a raw-yuv/h264 capable video grabber card. In raw mode, the source works fine without setting GST_BUFFER_OFFSET, GST_BUFFER_OFFSET_END, GST_BUFFER_TIMESTAMP and GST_BUFFER_DURATION, but for h264 live play, I can't find a working combination. The example pipeline looks like this: gst-launch mysource ! "video/x-h264,framerate=25/1" ! ffdec_h264 ! xvimagesink The card provides each frame as a separate buffer, and (in the current configuration) I have one SPS, one PPS, one I, and 14 P-frames in a group, each output in a separate GstBuffer; When not setting anything, the pipeline takes all grabbed frames, but displays only the first (or maybe first few). If I set all the values to what i believe is correct (put a serial number incrementing from 0 in OFFSET, OFFSET+1 in OFFSET_END, a hardware-generated timestamp in TIMESTAMP, and 0 for SPS/PPS frames and GST_SECOND/framerate for I/P frames in DURATION), the pipeline only takes and displays the first four frames and then stalls. If I count the SPS/PPS frames as normal frames, use the same duration for them as I/P frmaes and increment the timestamp accordingly, the buffer in my element fills slowly as the decoder takes fewer frames than produced. When saving the stream to a file and playing back from there, everything works fine. What would be the correct values for the timestamps in this case? Or do I have to implement a clock-capable element? -- Dani ...and Linux for all. From ensonic at hora-obscura.de Tue Sep 8 17:25:24 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Tue, 08 Sep 2009 18:25:24 +0300 Subject: [gst-devel] Info for codec_data parsing for H264 in a FLV container In-Reply-To: <400edb7e0909041031q40f71641t8b4b738b85d465ec@mail.gmail.com> References: <400edb7e0909041031q40f71641t8b4b738b85d465ec@mail.gmail.com> Message-ID: <4AA67764.3090302@hora-obscura.de> Joaquin Castellanos schrieb: > Hi > > I am looking for information required to parse H264 codec_data. > With some Flv containers (with H264 v-streams) the flvdemux does not > parse width, height or framerate, > instead the demuxer sent the codec_data to the next element. > > e.g. > > # gst-launch filesrc location > =/data/EVM_filesystems/x0089714/target/Vid.flv ! flvdemux name=demux > demux.video ! fakesink -v > Setting pipeline to PAUSED ... > /GstPipeline:pipeline0/GstFLVDemux:demux.GstPad:video: caps = > video/x-h264, pixel-aspect-ratio=(fraction)1/1, > codec_data=(buffer)01640033ffe1001c67640033ac2cc502d0ceffc01400144400000fa40003a9823c60c65801000468eebcb0 > > have a look at h264parse. Stefan > Best regards > Joaquin > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From ensonic at hora-obscura.de Tue Sep 8 17:26:13 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Tue, 08 Sep 2009 18:26:13 +0300 Subject: [gst-devel] Find out what sinks are avaiable. In-Reply-To: <25254429.post@talk.nabble.com> References: <25254429.post@talk.nabble.com> Message-ID: <4AA67795.90905@hora-obscura.de> 00a schrieb: > If I have several usb sound cards how do I select which one to output to? > I have installed gstreamer on my embedded system and can hear the tone > from by default headphone socket using > gst-launch-0.10 audiotestsrc ! audioconvert ! audioresample ! alsasink > I tried adding device=1 at the end but this did nothing apart from error. > Can someone help me out with the syntax and the ability to query what] > soundcards are available? > Its alsa names e.g. hw:0 or plughw:0. Stefan From ensonic at hora-obscura.de Tue Sep 8 17:27:30 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Tue, 08 Sep 2009 18:27:30 +0300 Subject: [gst-devel] ERROR: from element /pipeline0/timidity0: Could not decode stream. In-Reply-To: <6efe08af0909062350w31705eb7n1f1e58d4a27f6169@mail.gmail.com> References: <6efe08af0909060413u42207b0evff87f2a512e010f1@mail.gmail.com> <4AA40A53.4050409@hora-obscura.de> <6efe08af0909062350w31705eb7n1f1e58d4a27f6169@mail.gmail.com> Message-ID: <4AA677E2.9010308@hora-obscura.de> hi, Vitaly V. Ch schrieb: > This midi file played successfully via aplaymidi but on my system > gstreamer can't play midi files at all. > > I'm beginner in Gstreamer and in this case need guide. > well file a bug and attach the file. Then its easier to help. Stefan > \\wbr Vitaly > > On Sun, Sep 6, 2009 at 10:15 PM, Stefan Kost wrote: > >> Vitaly V. Ch schrieb: >> >>> I want to play midi via gstreamer but got next trouble: >>> >>> # LANG= gst-launch filesrc location=Krysha_doma_tvoego.mid ! timidity ! alsasink >>> Setting pipeline to PAUSED ... >>> Pipeline is PREROLLING ... >>> ERROR: from element /pipeline0/timidity0: Could not decode stream. >>> Additional debug info: >>> gsttimidity.c(641): gst_timidity_loop (): /pipeline0/timidity0: >>> Unable to parse midi >>> ERROR: pipeline doesn't want to preroll. >>> Setting pipeline to NULL ... >>> FREEING pipeline ... >>> # >>> >>> >> can you play that with other players. Please double check that its a midi-file >> (what does "file Krysha_doma_tvoego.mid" reports). If it only fails with >> gstreamer, please file a bug and attach the file. >> >> Stefan >> >> ------------------------------------------------------------------------------ >> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day >> trial. Simplify your report design, integration and deployment - and focus on >> what you do best, core application coding. Discover what's new with >> Crystal Reports now. http://p.sf.net/sfu/bobj-july >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From ensonic at hora-obscura.de Tue Sep 8 17:30:01 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Tue, 08 Sep 2009 18:30:01 +0300 Subject: [gst-devel] CPU Load bcoz of Gstreamer plugin In-Reply-To: <25343663.post@talk.nabble.com> References: <25340716.post@talk.nabble.com> <1252397608.2148.7.camel@putamadre> <25343663.post@talk.nabble.com> Message-ID: <4AA67879.9040209@hora-obscura.de> hi, rakesh sharma schrieb: > We dont have those tools callgrind/oprofile on the target platform. > I have written the plugin in lcml(socket node), so code should take maximum > cpu on dsp side not arm side, but its tacking full cpu on arm side . I am > not able to measure where its eating cpu. > If you people have those tools for arm you can share it with me so that i'll > check and let you know about that.please reply soon its urgent. > oprofile works on arm. Your kernel needs support for it. E.g. the maemo5 sdk has it. Stefan > thanks for reply... > > > Edward Hervey wrote: > >> Sorry to ask the obvious question... >> >> ... but have you profiled your pipeline and/or encoder to see *WHERE* >> it's taking 99% cpu ? Use a profiler like callgrind/oprofile (if >> available on your platform) or analyze your logs to see where it's >> taking all that time. >> >> Until you've analyzed where your problem is... we won't be able to >> help you. >> >> On Mon, 2009-09-07 at 23:47 -0700, rakesh sharma wrote: >> >>> I have written mp4sp encoder and its tacking 99% cpu load for all >>> resolutions. >>> Can anyone suggest solution for this ? >>> Is that i have to release cpu using some gstreamer api ? >>> Please help me its urgent..... >>> >> ------------------------------------------------------------------------------ >> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 >> 30-Day >> trial. Simplify your report design, integration and deployment - and focus >> on >> what you do best, core application coding. Discover what's new with >> Crystal Reports now. http://p.sf.net/sfu/bobj-july >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> > > From ensonic at hora-obscura.de Tue Sep 8 17:32:31 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Tue, 08 Sep 2009 18:32:31 +0300 Subject: [gst-devel] Help Running Gstreamer tests In-Reply-To: <25230217.post@talk.nabble.com> References: <25230217.post@talk.nabble.com> Message-ID: <4AA6790F.6010608@hora-obscura.de> tonybeck schrieb: > I can see there is a test dir in the gstreamer repository but do not know how > to execute these tests. Does anyone ever run them and can explain? Thanks. > run "make check". And yes, people are supposed to run them after making changes. Also try cd tests and make help for more options. Stefan From vitaly.v.ch at gmail.com Tue Sep 8 17:35:12 2009 From: vitaly.v.ch at gmail.com (Vitaly V. Ch) Date: Tue, 8 Sep 2009 18:35:12 +0300 Subject: [gst-devel] ERROR: from element /pipeline0/timidity0: Could not decode stream. In-Reply-To: <4AA677E2.9010308@hora-obscura.de> References: <6efe08af0909060413u42207b0evff87f2a512e010f1@mail.gmail.com> <4AA40A53.4050409@hora-obscura.de> <6efe08af0909062350w31705eb7n1f1e58d4a27f6169@mail.gmail.com> <4AA677E2.9010308@hora-obscura.de> Message-ID: <6efe08af0909080835t7cccbb54md88e630604dcd330@mail.gmail.com> Ok, I will do it after tomorrow day. On Tue, Sep 8, 2009 at 6:27 PM, Stefan Kost wrote: > hi, > Vitaly V. Ch schrieb: >> This midi file played successfully via aplaymidi but on my system >> gstreamer can't play midi files at all. >> >> I'm beginner in Gstreamer and in this case need guide. >> > well file a bug and attach the file. Then its easier to help. > > Stefan > >> \\wbr Vitaly >> >> On Sun, Sep 6, 2009 at 10:15 PM, Stefan Kost wrote: >> >>> Vitaly V. Ch schrieb: >>> >>>> I want to play midi via gstreamer but got next trouble: >>>> >>>> # LANG= gst-launch filesrc location=Krysha_doma_tvoego.mid ! timidity ! alsasink >>>> Setting pipeline to PAUSED ... >>>> Pipeline is PREROLLING ... >>>> ERROR: from element /pipeline0/timidity0: Could not decode stream. >>>> Additional debug info: >>>> gsttimidity.c(641): gst_timidity_loop (): /pipeline0/timidity0: >>>> Unable to parse midi >>>> ERROR: pipeline doesn't want to preroll. >>>> Setting pipeline to NULL ... >>>> FREEING pipeline ... >>>> # >>>> >>>> >>> can you play that with other players. Please double check that its a midi-file >>> (what does "file Krysha_doma_tvoego.mid" reports). If it only fails with >>> gstreamer, please file a bug and attach the file. >>> >>> Stefan >>> >>> ------------------------------------------------------------------------------ >>> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day >>> trial. Simplify your report design, integration and deployment - and focus on >>> what you do best, core application coding. Discover what's new with >>> Crystal Reports now. ?http://p.sf.net/sfu/bobj-july >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> >> ------------------------------------------------------------------------------ >> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day >> trial. Simplify your report design, integration and deployment - and focus on >> what you do best, core application coding. Discover what's new with >> Crystal Reports now. ?http://p.sf.net/sfu/bobj-july >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. ?http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From vitaly.v.ch at gmail.com Tue Sep 8 17:36:18 2009 From: vitaly.v.ch at gmail.com (Vitaly V. Ch) Date: Tue, 8 Sep 2009 18:36:18 +0300 Subject: [gst-devel] ERROR: from element /pipeline0/timidity0: Could not decode stream. In-Reply-To: <4AA677E2.9010308@hora-obscura.de> References: <6efe08af0909060413u42207b0evff87f2a512e010f1@mail.gmail.com> <4AA40A53.4050409@hora-obscura.de> <6efe08af0909062350w31705eb7n1f1e58d4a27f6169@mail.gmail.com> <4AA677E2.9010308@hora-obscura.de> Message-ID: <6efe08af0909080836k2a2439bbr888a669cc0ea9584@mail.gmail.com> Ok, I will do it tomorrow. On Tue, Sep 8, 2009 at 6:27 PM, Stefan Kost wrote: > hi, > Vitaly V. Ch schrieb: >> This midi file played successfully via aplaymidi but on my system >> gstreamer can't play midi files at all. >> >> I'm beginner in Gstreamer and in this case need guide. >> > well file a bug and attach the file. Then its easier to help. > > Stefan > >> \\wbr Vitaly >> >> On Sun, Sep 6, 2009 at 10:15 PM, Stefan Kost wrote: >> >>> Vitaly V. Ch schrieb: >>> >>>> I want to play midi via gstreamer but got next trouble: >>>> >>>> # LANG= gst-launch filesrc location=Krysha_doma_tvoego.mid ! timidity ! alsasink >>>> Setting pipeline to PAUSED ... >>>> Pipeline is PREROLLING ... >>>> ERROR: from element /pipeline0/timidity0: Could not decode stream. >>>> Additional debug info: >>>> gsttimidity.c(641): gst_timidity_loop (): /pipeline0/timidity0: >>>> Unable to parse midi >>>> ERROR: pipeline doesn't want to preroll. >>>> Setting pipeline to NULL ... >>>> FREEING pipeline ... >>>> # >>>> >>>> >>> can you play that with other players. Please double check that its a midi-file >>> (what does "file Krysha_doma_tvoego.mid" reports). If it only fails with >>> gstreamer, please file a bug and attach the file. >>> >>> Stefan >>> >>> ------------------------------------------------------------------------------ >>> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day >>> trial. Simplify your report design, integration and deployment - and focus on >>> what you do best, core application coding. Discover what's new with >>> Crystal Reports now. ?http://p.sf.net/sfu/bobj-july >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> >> ------------------------------------------------------------------------------ >> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day >> trial. Simplify your report design, integration and deployment - and focus on >> what you do best, core application coding. Discover what's new with >> Crystal Reports now. ?http://p.sf.net/sfu/bobj-july >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. ?http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From tristan at sat.qc.ca Tue Sep 8 17:39:37 2009 From: tristan at sat.qc.ca (Tristan Matthews) Date: Tue, 08 Sep 2009 11:39:37 -0400 Subject: [gst-devel] udpsrc multicast In-Reply-To: <25314433.post@talk.nabble.com> References: <3afe75670907211035kb4871abl76f7bb4a3d67ca57@mail.gmail.com> <25314433.post@talk.nabble.com> Message-ID: <4AA67AB9.5060403@sat.qc.ca> have you tried specifying the multicast-iface property? -Tristan attaboy wrote: > udpsrc is broken on Windows. There is a patch but I don't think it has made > it into the trunk yet. > > http://bugzilla.gnome.org/show_bug.cgi?id=534243 > > > > Levi Pope wrote: > >> I am trying to use udpsrc on windows to pull in a stream on a multicast >> address >> but all I get is this error. >> >> c:\gst-launch-0.10 --gst-debug=udpsrc:4 udpsrc multicast-group=224.1.2.3 >> port=1234 ! fakesink dump=1 >> >> 0:00:00.093750000 4968 00352940 DEBUG udpsrc >> gstudpsrc.c:611:gst_udpsrc_update_uri: updated uri to udp:// >> 224.1.2.3 >> :4951 >> 0:00:00.093750000 4968 00352940 DEBUG udpsrc >> gstudpsrc.c:611:gst_udpsrc_update_uri: updated uri to udp:// >> 224.1.2.3 >> :1234 >> Setting pipeline to PAUSED ... >> 0:00:00.125000000 4968 00352940 DEBUG udpsrc >> gstudpsrc.c:806:gst_udpsrc_start: allocating socket for >> 224.1.2.3:123 >> 4 >> 0:00:00.125000000 4968 00352940 DEBUG udpsrc >> gstudpsrc.c:825:gst_udpsrc_start: binding on port 1234 >> 0:00:00.125000000 4968 00352940 WARN udpsrc >> gstudpsrc.c:967:gst_udpsrc_start: error: bind failed -1: No error >> (0) >> ERROR: Pipeline doesn't want to pause. >> ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Could not >> get/set settings from/on resource. >> Additional debug info: >> ..\..\..\Source\gst-plugins-good\gst\udp\gstudpsrc.c(967): >> gst_udpsrc_start >> (): /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: >> bind failed -1: No error (0) >> Setting pipeline to NULL ... >> >> Is there something I am missing? >> >> Thanks for the help >> Levi >> >> ------------------------------------------------------------------------------ >> >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> > > -- Tristan Matthews Soci?t? des arts technologiques [SAT] email: tristan at sat.qc.ca web: http://www.music.mcgill.ca/~tmatthews From smcnam at gmail.com Tue Sep 8 17:40:44 2009 From: smcnam at gmail.com (Sean McNamara) Date: Tue, 8 Sep 2009 11:40:44 -0400 Subject: [gst-devel] how to cleanly re-install gstreamer? Help!! In-Reply-To: <25252421.post@talk.nabble.com> References: <25252421.post@talk.nabble.com> Message-ID: <74eb1fe20909080840i6606a0aaob90ba78e75a525cd@mail.gmail.com> Generally it's not a good idea to switch out a different version of the GStreamer core under the feet of installed plugins. When you do that, you should rebuild the plugins. I'm assuming you are on Linux since you use Synaptic Package Manager. You have two options to recover a working gstreamer installation: 1. Reinstall all gstreamer packages from the package manager. You will lose the ability to use gst-rtsp-server and your compiled gstreamer core will be overwritten with what's in the package manager. Figuring out the names of every gstreamer package in order to reinstall them is a question you should ask to your Linux distribution support channel. 2. Compile all the desired gstreamer plugins from source. This will overwrite any plugins with the same names from the package manager. Also, if you ever allow your package manager to apply a software "update" to your gstreamer packages, it will happily overwrite your work. To prevent that, you may want to place a "hold" on these packages to indicate they should not be upgraded. Again, a question for your distribution support. For this option you'll also need the development packages of the plugins' many dependencies. Depending on how thorough your distribution is, this method could take a lot of work. GStreamer is just a set of files -- shared libraries, mostly. Treat it as such. A "clean install", intuitively, consists of deleting all the current files and installing new ones to replace them. If you use your package manager, it does that automatically and lets you think of it in terms of "installed" and "uninstalled" (an unnecessary abstraction for this case). If you compile from source, you're just overwriting the files that the package manager claims to have "installed". -Sean On Sat, Sep 5, 2009 at 5:06 PM, Yiliang Bao wrote: > > Hi, > > I compiled and installed the latest gst-rtsp-server, and updated gstreamer > itself based on the error messages from gst-rtsp-server installation. > However, after these steps, I could no longer find some plugins like mad, > xvimagesink, v4l2src, etc. > > I have tried re-installing the latest versions of all plugin packages. I > have also tried to remove, then re-install all plugins from Synaptic Package > Manager. Got some plugin, like v4l2src, back, but mad, xvimagesink are still > missing. autovideosink is there, but some pipleine which works is no longer > working. For example: > > gst-launch v4l2src device=/dev/video0 ! autovideosink > >>>>>>> Error messages: > > Setting pipeline to PAUSED ... > ERROR: Pipeline doesn't want to pause. > ERROR: from element /GstPipeline:pipeline0/GstAutoVideoSink:autovideosink0: > Could not initialize supporting library. > Additional debug info: > gstautovideosink.c(373): gst_auto_video_sink_detect (): > /GstPipeline:pipeline0/GstAutoVideoSink:autovideosink0: > Failed to set target pad > Setting pipeline to NULL ... > Freeing pipeline ... > > gst-launch filesrc location=Videos/xyz.avi ! decodebin ! autovideosink > >>>>>>>>>> Error messages: > > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > ERROR: from element > /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstAviDemux:avidemux0: > Internal data stream error. > Additional debug info: > gstavidemux.c(4443): gst_avi_demux_loop (): > /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstAviDemux:avidemux0: > streaming stopped, reason not-linked > ERROR: pipeline doesn't want to preroll. > Setting pipeline to NULL ... > Freeing pipeline ... > > Does anyone know why? Is it possible to re-install a clean version of > gstreamer? > > Yiliang > -- > View this message in context: http://www.nabble.com/how-to-cleanly-re-install-gstreamer--Help%21%21-tp25252421p25252421.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. ?http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From julien.isorce at gmail.com Tue Sep 8 17:53:38 2009 From: julien.isorce at gmail.com (Julien Isorce) Date: Tue, 8 Sep 2009 17:53:38 +0200 Subject: [gst-devel] about capturing 5.1 and 7.1 sources Message-ID: <180a127d0909080853v869d1f2ie3efc772ed7aac3b@mail.gmail.com> hi, There is a ac3iec958 element that can convert a ac" stream to raw int stereo data (audio/x-raw-int, channels=2). Am I correct ? Well, Is there a rawIntStereo_to_ac3 element ? does it make sense ? (Because I heard tha ac3 can be provided over raw stereo.) Does anyone can share its experience about capturing 5.1 or 7.1 sources ? (from blueray or any kind of sources) Maybe there is somewhere a copyright limitation, I mean something like HDCP .... Sincerely Julien -------------- next part -------------- An HTML attachment was scrubbed... URL: From smcnam at gmail.com Tue Sep 8 17:54:19 2009 From: smcnam at gmail.com (Sean McNamara) Date: Tue, 8 Sep 2009 11:54:19 -0400 Subject: [gst-devel] new to gstreamer In-Reply-To: <25288295.post@talk.nabble.com> References: <25288295.post@talk.nabble.com> Message-ID: <74eb1fe20909080854s6289757eo8c678d405b3ca377@mail.gmail.com> Hi, On Fri, Sep 4, 2009 at 12:54 AM, co.sam wrote: > > Hi friends, > Hope you are doing great.Recently i started reading gstremer to play around > with media files but I am absolutely new to gstreamer. > > I refered an application manual of gsteamer.Also i tried some gst-launch > examples such as: > > 1.gst-launch audiotestsrc ! audioconvert ! audioresample ! osssink(plays > sound) > > 2.gst-launch-0.10 ximagesrc num-buffers=1 ! ffmpegcolorspace ! pngenc ! > filesink location=screenshot.png(captures the screen shot) > > Though I have tried these examples I have not understood the exact meaning > of what these individual elements of this pipeline are donig and also how > come these pipelines are constructed in this particular order. > > Please help in out to learn gstremer > Any kind of help is welcome....... These are very general questions that can be answered either by reading the documentation ( http://gstreamer.freedesktop.org/documentation/ ) or by reading the source code. Each plugin and element class has documentation linked to from that same page as well. The pipeline is constructed in that order because all data flows from a source to a sink. That's just the way gstreamer (and most other filter graph based media pipelines) are designed. Data originates from a certain element; the data is processed; and the output data flows to one or more sinks. Thus, there are at least three basic categories of elements: sources, sinks, and the rest. Sources only "spit out" data; they don't require you to send them any data. They get their data from somewhere else. For instance, audiotestsrc generates PCM data based on sinusoidal wave patterns that can be constructed mathematically. Sinks only "consume" data; they don't have the ability to send data to any other element. Sinks always get their data from somewhere else, and deliver it to some end user. For instance, filesink consumes data and writes it out to a file on disk. The rest of the elements both consume and spit out data. To construct a valid pipeline, any element that is neither a source nor a sink must be "between" two other elements: one on the left, and one on the right. The gst-launch pipeline syntax is designed to visualize data flow progressing from left to right. On the left is the source and on the right is the sink. You can think of it as right to left if you want, but the pipeline syntax parser probably won't like that. If you're creating elements programmatically, feel free to create the sink first, and the source last. Also note that pipelines needn't be perfectly linear; you can have data originating from one source that ends up feeding multiple sinks. And of course you can sometimes have data from multiple sources coalesce into one sink. There are elements for most types of data that go both ways. HTH, -Sean > > regards, > co.sam > -- > View this message in context: http://www.nabble.com/new-to-gstreamer-tp25288295p25288295.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. ?http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From julien.isorce at gmail.com Tue Sep 8 17:56:43 2009 From: julien.isorce at gmail.com (Julien Isorce) Date: Tue, 8 Sep 2009 17:56:43 +0200 Subject: [gst-devel] Problems with AV Sync with Open GL sink in Windows In-Reply-To: <25335761.post@talk.nabble.com> References: <25335761.post@talk.nabble.com> Message-ID: <180a127d0909080856m4480c90bm9ea05571b1848957@mail.gmail.com> 2009/9/7 Mark Sauer > > > The problem I am having with glimagesink is that the audio and video tracks > are not in sync. I am wondering if anyone has managed to get glimagesink > to > work on Windows with the AV in sync, and if so how this was done. And if > What is your pipeline ? a link on the file ? in order to reproduce the problem. Do you mean, you got an audio/video file somewhere that is ok on linux with gstreamer and not on windows with glimagesink ? or ok on windows with and other player ? Julien > > Thanks for any help. > Mark Sauer > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From wim.taymans at gmail.com Tue Sep 8 18:03:56 2009 From: wim.taymans at gmail.com (Wim Taymans) Date: Tue, 08 Sep 2009 18:03:56 +0200 Subject: [gst-devel] about capturing 5.1 and 7.1 sources In-Reply-To: <180a127d0909080853v869d1f2ie3efc772ed7aac3b@mail.gmail.com> References: <180a127d0909080853v869d1f2ie3efc772ed7aac3b@mail.gmail.com> Message-ID: <1252425836.11128.22.camel@metal> On Tue, 2009-09-08 at 17:53 +0200, Julien Isorce wrote: > hi, > > There is a ac3iec958 element that can convert a ac" stream to raw int > stereo data (audio/x-raw-int, channels=2). Am I correct ? No, ac3iec958 pads AC3 frames into IEC958 frames suitable for a raw S/PDIF interface. It does not convert anything to raw audio. > > Well, Is there a rawIntStereo_to_ac3 element ? does it make sense ? > (Because I heard tha ac3 can be provided over raw stereo.) audio/x-raw-int to ac3 would be ffenc_ac3.. Wim > > Does anyone can share its experience about capturing 5.1 or 7.1 > sources ? (from blueray or any kind of sources) > > Maybe there is somewhere a copyright limitation, I mean something like > HDCP .... > > Sincerely > > Julien > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From julien.isorce at gmail.com Tue Sep 8 18:14:00 2009 From: julien.isorce at gmail.com (Julien Isorce) Date: Tue, 8 Sep 2009 18:14:00 +0200 Subject: [gst-devel] about capturing 5.1 and 7.1 sources In-Reply-To: <1252425836.11128.22.camel@metal> References: <180a127d0909080853v869d1f2ie3efc772ed7aac3b@mail.gmail.com> <1252425836.11128.22.camel@metal> Message-ID: <180a127d0909080914r6161029eme6e2e881a262d69f@mail.gmail.com> 2009/9/8 Wim Taymans > On Tue, 2009-09-08 at 17:53 +0200, Julien Isorce wrote: > > hi, > > > > There is a ac3iec958 element that can convert a ac" stream to raw int > > stereo data (audio/x-raw-int, channels=2). Am I correct ? > > No, ac3iec958 pads AC3 frames into IEC958 frames suitable for a raw > S/PDIF interface. It does not convert anything to raw audio. > This is what I mean: ac3 over raw stereo. I did not talk about decoding like ffdec_ac3 Sorry i was not very precise: 'raw-audio' property on ac3iec958 element > > > > > Well, Is there a rawIntStereo_to_ac3 element ? does it make sense ? > > (Because I heard tha ac3 can be provided over raw stereo.) > > audio/x-raw-int to ac3 would be ffenc_ac3.. > ok there is no iec958ac3 so. Thx Julien > > Wim > > > > Does anyone can share its experience about capturing 5.1 or 7.1 > > sources ? (from blueray or any kind of sources) > > > > Maybe there is somewhere a copyright limitation, I mean something like > > HDCP .... > > > > Sincerely > > > > Julien > > > ------------------------------------------------------------------------------ > > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 > 30-Day > > trial. Simplify your report design, integration and deployment - and > focus on > > what you do best, core application coding. Discover what's new with > > Crystal Reports now. http://p.sf.net/sfu/bobj-july > > _______________________________________________ gstreamer-devel mailing > list gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus > on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From lrn1986 at gmail.com Tue Sep 8 18:23:34 2009 From: lrn1986 at gmail.com (LRN) Date: Tue, 08 Sep 2009 20:23:34 +0400 Subject: [gst-devel] how to cleanly re-install gstreamer? Help!! In-Reply-To: <74eb1fe20909080840i6606a0aaob90ba78e75a525cd@mail.gmail.com> References: <25252421.post@talk.nabble.com> <74eb1fe20909080840i6606a0aaob90ba78e75a525cd@mail.gmail.com> Message-ID: <4AA68506.2030603@gmail.com> Sean McNamara wrote: > Generally it's not a good idea to switch out a different version of > the GStreamer core under the feet of installed plugins. When you do > that, you should rebuild the plugins. > > I'm assuming you are on Linux since you use Synaptic Package Manager. > > You have two options to recover a working gstreamer installation: > > 1. Reinstall all gstreamer packages from the package manager. You will > lose the ability to use gst-rtsp-server and your compiled gstreamer > core will be overwritten with what's in the package manager. Figuring > out the names of every gstreamer package in order to reinstall them is > a question you should ask to your Linux distribution support channel. > > 2. Compile all the desired gstreamer plugins from source. This will > overwrite any plugins with the same names from the package manager. > Also, if you ever allow your package manager to apply a software > "update" to your gstreamer packages, it will happily overwrite your > work. To prevent that, you may want to place a "hold" on these > packages to indicate they should not be upgraded. Again, a question > for your distribution support. For this option you'll also need the > development packages of the plugins' many dependencies. Depending on > how thorough your distribution is, this method could take a lot of > work. > > GStreamer is just a set of files -- shared libraries, mostly. Treat it > as such. A "clean install", intuitively, consists of deleting all the > current files and installing new ones to replace them. If you use your > package manager, it does that automatically and lets you think of it > in terms of "installed" and "uninstalled" (an unnecessary abstraction > for this case). If you compile from source, you're just overwriting > the files that the package manager claims to have "installed". > > -Sean > > On Sat, Sep 5, 2009 at 5:06 PM, Yiliang Bao wrote: > >> Hi, >> >> I compiled and installed the latest gst-rtsp-server, and updated gstreamer >> itself based on the error messages from gst-rtsp-server installation. >> However, after these steps, I could no longer find some plugins like mad, >> xvimagesink, v4l2src, etc. >> >> I have tried re-installing the latest versions of all plugin packages. I >> have also tried to remove, then re-install all plugins from Synaptic Package >> Manager. Got some plugin, like v4l2src, back, but mad, xvimagesink are still >> missing. autovideosink is there, but some pipleine which works is no longer >> working. For example: >> >> gst-launch v4l2src device=/dev/video0 ! autovideosink >> >> >>>>>>>> Error messages: >>>>>>>> >> Setting pipeline to PAUSED ... >> ERROR: Pipeline doesn't want to pause. >> ERROR: from element /GstPipeline:pipeline0/GstAutoVideoSink:autovideosink0: >> Could not initialize supporting library. >> Additional debug info: >> gstautovideosink.c(373): gst_auto_video_sink_detect (): >> /GstPipeline:pipeline0/GstAutoVideoSink:autovideosink0: >> Failed to set target pad >> Setting pipeline to NULL ... >> Freeing pipeline ... >> >> gst-launch filesrc location=Videos/xyz.avi ! decodebin ! autovideosink >> >> >>>>>>>>>>> Error messages: >>>>>>>>>>> >> Setting pipeline to PAUSED ... >> Pipeline is PREROLLING ... >> ERROR: from element >> /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstAviDemux:avidemux0: >> Internal data stream error. >> Additional debug info: >> gstavidemux.c(4443): gst_avi_demux_loop (): >> /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstAviDemux:avidemux0: >> streaming stopped, reason not-linked >> ERROR: pipeline doesn't want to preroll. >> Setting pipeline to NULL ... >> Freeing pipeline ... >> >> Does anyone know why? Is it possible to re-install a clean version of >> gstreamer? >> >> Yiliang >> -- >> View this message in context: http://www.nabble.com/how-to-cleanly-re-install-gstreamer--Help%21%21-tp25252421p25252421.html >> Sent from the GStreamer-devel mailing list archive at Nabble.com. >> Also, it may be worthwhile to erase your plugin cache and let GStreamer regenerate it. From t.i.m at zen.co.uk Tue Sep 8 18:58:28 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Tue, 08 Sep 2009 17:58:28 +0100 Subject: [gst-devel] about capturing 5.1 and 7.1 sources In-Reply-To: <180a127d0909080853v869d1f2ie3efc772ed7aac3b@mail.gmail.com> References: <180a127d0909080853v869d1f2ie3efc772ed7aac3b@mail.gmail.com> Message-ID: <1252429108.5124.33.camel@zingle> On Tue, 2009-09-08 at 17:53 +0200, Julien Isorce wrote: > There is a ac3iec958 element that can convert a ac" stream to raw int > stereo data (audio/x-raw-int, channels=2). Am I correct ? > > Well, Is there a rawIntStereo_to_ac3 element ? does it make sense ? > (Because I heard that ac3 can be provided over raw stereo.) Some people put AC3 or DTS data into 16-bit stereo PCM streams, but this is mostly a "hack" as far as I know. This kind of mis-labelling of the stream contents should not really be needed. I don't think there's an iec958ac3 element yet. Alsasrc would probably need a few fixes before this would work anyway. Cheers -Tim From gstelzz at yahoo.fr Tue Sep 8 21:42:09 2009 From: gstelzz at yahoo.fr (Aurelien Grimaud) Date: Tue, 08 Sep 2009 21:42:09 +0200 Subject: [gst-devel] What does this GStreamer error mean? In-Reply-To: <92f1bb260909080614t433d3208wbac0162f6637ee1d@mail.gmail.com> References: <92f1bb260909080614t433d3208wbac0162f6637ee1d@mail.gmail.com> Message-ID: <4AA6B391.2010608@yahoo.fr> Well, > man pthread_mutex_destroy > ERRORS > The pthread_mutex_destroy() function may fail if: > > EBUSY The implementation has detected an attempt to destroy > the object referenced by mutex while it is locked > or referenced (for example, while being used in a > pthread_cond_timedwait() or pthread_cond_wait()) by > another thread. Last reference on your udpsrc is dropped while it is still running. One of the mutex of udpsrc is locked, while being destroyed in the dispose function of udpsrc. As said : you need to explicitely set the udpsrc to NULL state before dropping the final reference. If you already set it to NULL before removing from pipeline, this means you have a refcount problem somewhere. Aurelien Le 08/09/2009 15:14, Marl?ne Hildebrand-Ehrhardt a ?crit : > Hi! > > I'm using a streaming application that has been developped with Java > and GStreamer. > Sometimes, when I launch the application, after a few streams have > been created, the application crashes and I get the following message : > > [java] > [java] (JavaUBIK:23929): GStreamer-CRITICAL **: > [java] Trying to dispose element udpsrc34, but it is not in the > NULL state. > [java] You need to explicitly set elements to the NULL state before > [java] dropping the final reference, to allow them to clean up. > [java] > [java] > [java] GThread-ERROR **: file > /build/buildd/glib2.0-2.20.1/gthread/gthread-posix.c: line 171 > (g_mutex_free_posix_impl): error 'P?riph?rique ou ressource occup?' > during 'pthread_mutex_destroy ((pthread_mutex_t *) mutex)' > [java] aborting... > [java] Java Result: 134 > > Does anybody have an idea of the signification of this message? > > Thank you by advance! > > Marl?ne Hildebrand-Ehrhardt > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ds at entropywave.com Tue Sep 8 22:44:29 2009 From: ds at entropywave.com (David Schleef) Date: Tue, 8 Sep 2009 13:44:29 -0700 Subject: [gst-devel] how to cleanly re-install gstreamer? Help!! In-Reply-To: <74eb1fe20909080840i6606a0aaob90ba78e75a525cd@mail.gmail.com> References: <25252421.post@talk.nabble.com> <74eb1fe20909080840i6606a0aaob90ba78e75a525cd@mail.gmail.com> Message-ID: <20090908204429.GA10868@entropywave.com> On Tue, Sep 08, 2009 at 11:40:44AM -0400, Sean McNamara wrote: > Generally it's not a good idea to switch out a different version of > the GStreamer core under the feet of installed plugins. This is incorrect. As long as both core versions satisfy the requirements of the plugins, you should be fine. dave... From julien.isorce at gmail.com Wed Sep 9 01:23:35 2009 From: julien.isorce at gmail.com (Julien Isorce) Date: Wed, 9 Sep 2009 01:23:35 +0200 Subject: [gst-devel] about capturing 5.1 and 7.1 sources In-Reply-To: <1252429108.5124.33.camel@zingle> References: <180a127d0909080853v869d1f2ie3efc772ed7aac3b@mail.gmail.com> <1252429108.5124.33.camel@zingle> Message-ID: <180a127d0909081623v19a27c69y1f339a6454202650@mail.gmail.com> 2009/9/8 Tim-Philipp M?ller > On Tue, 2009-09-08 at 17:53 +0200, Julien Isorce wrote: > > > There is a ac3iec958 element that can convert a ac" stream to raw int > > stereo data (audio/x-raw-int, channels=2). Am I correct ? > > > > Well, Is there a rawIntStereo_to_ac3 element ? does it make sense ? > > (Because I heard that ac3 can be provided over raw stereo.) > > Some people put AC3 or DTS data into 16-bit stereo PCM streams, but this > is mostly a "hack" as far as I know. This kind of mis-labelling of the > stream contents should not really be needed. > ok > > I don't think there's an iec958ac3 element yet. ok > Alsasrc would probably > need a few fixes before this would work anyway. > Why ? I think spdif input also supports 16 bits stereo raw int. Well It seems that newer capture devices are responsible for decoding ac3 or dts, in hardware. And so directly provide pcm. I am currently testing the sound blaster X-Fi extreme audio, but I can't capture and resitute 5.1 or 7.1. (linux: driver pb, seems to be not correctly reconized by alsa) win: maximun 2 channels) If anyone already succedeed to capture 5.1 or 7.1 with a correct restitution, it would be appreciated to share the model of the capture card. (linux or other os) Sincerely Julien > Cheers > -Tim > > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus > on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jaisena at yahoo.com Wed Sep 9 06:43:36 2009 From: jaisena at yahoo.com (jayasena s) Date: Tue, 8 Sep 2009 21:43:36 -0700 (PDT) Subject: [gst-devel] A/V sync and stutter issue Message-ID: <316913.58035.qm@web55304.mail.re4.yahoo.com> Hi, ?I am having A/V Sync and stutter issues , when using totem player with gstreamer. With gst-launch, audio and video are in sync and video is smooth ( no stutter issue). ? Timestamps seem to be good, the stutter is more visible for the first 4 seconds of the playback. ? Did anyone come across this issue, Could you suggest ways to debug these issues ? Thanks, Jai -------------- next part -------------- An HTML attachment was scrubbed... URL: From rana.manishrana at gmail.com Wed Sep 9 07:58:39 2009 From: rana.manishrana at gmail.com (Manish Rana) Date: Wed, 9 Sep 2009 11:28:39 +0530 Subject: [gst-devel] Fwd: [FFmpeg-devel] debug info In-Reply-To: <49634284-D964-493D-A716-C68E9C3AF8FE@gmail.com> References: <49634284-D964-493D-A716-C68E9C3AF8FE@gmail.com> Message-ID: ask him to user g_print..or fprintf(stderr,.......) On Sun, Aug 30, 2009 at 8:14 PM, Ronald S. Bultje wrote: > Hi, > > Can someone help this guy? > > Cheers, > Ronald > > Begin forwarded message: > > *From:* kkumar s > *Date:* August 29, 2009 11:01:30 PM EDT > *To:* ffmpeg-devel at mplayerhq.hu > *Subject:* *[FFmpeg-devel] debug info* > *Reply-To:* FFmpeg development discussions and patches < > ffmpeg-devel at mplayerhq.hu> > > Hi, > I'm a newbie to ffmpeg . I am trying to used gstreamer ffmeg plugin for > my console based application and I am trying to debug using printfs but I > get error / warnings > when I compile with printf. I tried tprintf and av_log , but I am not > able > to see where the error or log messages are located ? > Can you please let me know where I can see the messages or any general way > to debug using printfs ? > thanks, > kumar. > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel at mplayerhq.hu > https://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-devel > > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus > on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From fthiery at gmail.com Wed Sep 9 08:18:17 2009 From: fthiery at gmail.com (Florent) Date: Wed, 9 Sep 2009 08:18:17 +0200 Subject: [gst-devel] Naming threads ? Other performance debugging helpers ? Message-ID: <1efe3a6e0909082318i1581b7dt3e9afce6ec6620ce@mail.gmail.com> Hi I was wondering if it would be feasible that elements that launch threads (e.g. queues) could name threads (using prctl(PR_SET_NAME? ), with the name being constructed after their name, or the neighbour elements' name. After a quick search [1], i'm not sure the Linux kernel offers this feature This would help a lot performance debugging : so far if you want to know what thread is consuming the most power, you make assumptions on the nature of processing, and study partial pipelines to estimate the proportion taken by what, but this approach is very limiting for detecting/understanding side effects. Also, any other idea about how to "map" the CPU cycles consumption of a pipeline or other performance indicators would be welcome. I am considering developing a tool which would display states/hints of a pipeline's behaviour in a graphical fashion. So far i experimented with polling queue filling states, but it does not always prove helpful. I also don't know how i can access QoS data from python. Regards, FLorent [1] http://www.gossamer-threads.com/lists/engine?list=linux&do=search_results&search_forum=forum_1&search_string=thread+name&search_type=AND From bilboed at gmail.com Wed Sep 9 08:36:22 2009 From: bilboed at gmail.com (Edward Hervey) Date: Wed, 09 Sep 2009 08:36:22 +0200 Subject: [gst-devel] Fwd: [FFmpeg-devel] debug info In-Reply-To: References: <49634284-D964-493D-A716-C68E9C3AF8FE@gmail.com> Message-ID: <1252478182.2525.1.camel@putamadre> All av_log calls from ffmpeg are forwarded through gst-ffmpeg. Just use the 'ffmpeg' gstreamer debugging category to view them. GST_DEBUG=ffmpeg:5 gst-launc-0.10 blah blah blah Edward On Wed, 2009-09-09 at 11:28 +0530, Manish Rana wrote: > ask him to user g_print..or fprintf(stderr,.......) > > On Sun, Aug 30, 2009 at 8:14 PM, Ronald S. Bultje > wrote: > Hi, > > > Can someone help this guy? > > > Cheers, > Ronald > > Begin forwarded message: > > > > From: kkumar s > > Date: August 29, 2009 11:01:30 PM EDT > > To: ffmpeg-devel at mplayerhq.hu > > Subject: [FFmpeg-devel] debug info > > Reply-To: FFmpeg development discussions and patches > > > > > > > > > Hi, > > I'm a newbie to ffmpeg . I am trying to used gstreamer > > ffmeg plugin for > > my console based application and I am trying to debug using > > printfs but I > > get error / warnings > > when I compile with printf. I tried tprintf and av_log , > > but I am not able > > to see where the error or log messages are located ? > > Can you please let me know where I can see the messages or > > any general way > > to debug using printfs ? > > thanks, > > kumar. > > _______________________________________________ > > ffmpeg-devel mailing list > > ffmpeg-devel at mplayerhq.hu > > https://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-devel > > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal > Reports 2008 30-Day > trial. Simplify your report design, integration and deployment > - and focus on > what you do best, core application coding. Discover what's new > with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From yiliangb at gmail.com Wed Sep 9 08:59:07 2009 From: yiliangb at gmail.com (Yiliang Bao) Date: Tue, 8 Sep 2009 23:59:07 -0700 (PDT) Subject: [gst-devel] how to cleanly re-install gstreamer? Help!! In-Reply-To: <4AA68506.2030603@gmail.com> References: <25252421.post@talk.nabble.com> <74eb1fe20909080840i6606a0aaob90ba78e75a525cd@mail.gmail.com> <4AA68506.2030603@gmail.com> Message-ID: <25359622.post@talk.nabble.com> Hi Sean, Thanks a lot for the detailed explanation. I tried both options. Here is what I encountered. Some of information might be helpful for other people who gets into the same trouble. What I did --> Originally I installed all the packages using "apt-get ****" from command line. So they should be official releases. I upgraded gstreamer by compiling the latest source, it worked fine after resolving some library issues. After I ran ./configure for the latest base plugin package before I ran make and install, I found that all the old plugins (almost) were gone. I was able to resolve most dependencies by installing the development libraries (libXXX-dev) except for 4 plugins related to header files. After I installed the base set, I moved on to the good set, and found more problems. I patiently tried to solve all the library related dependency issues, but was left with more plugins that could not be built because of header file related issues. After I finished upgrading base and good set. I tried to construct some pipelines with plugins already installed after my best efforts mentioned above, but no luck. These pipelines that were working previously no longer worked. 1. Reinstall the old packages from package manager After I messed up with gstreamer, I tried to install the old packages using "apt-get" command, but it said all the packages are the latest, even though I knew most plugins were already removed. So this option seems to have some problem. 2. Continue compiling from source After wasting quite some time, I decided to move ahead with other packages by ignoring these plugins that I had no idea how to resolve the issues. Resolving library related dependency issues was rather tedious, but it is fairly straightforward. I just tried to find the development libraries in package manager with name similar to that complained in the ./configure output. Finally, I was able to able to install all the packages. Although many plugins were left un-installed, gstreamer works fine for the functions I tested, and RTSP server is also working now. Thanks again for all the replies! Yiliang L.R.N wrote: > > Sean McNamara wrote: >> Generally it's not a good idea to switch out a different version of >> the GStreamer core under the feet of installed plugins. When you do >> that, you should rebuild the plugins. >> >> I'm assuming you are on Linux since you use Synaptic Package Manager. >> >> You have two options to recover a working gstreamer installation: >> >> 1. Reinstall all gstreamer packages from the package manager. You will >> lose the ability to use gst-rtsp-server and your compiled gstreamer >> core will be overwritten with what's in the package manager. Figuring >> out the names of every gstreamer package in order to reinstall them is >> a question you should ask to your Linux distribution support channel. >> >> 2. Compile all the desired gstreamer plugins from source. This will >> overwrite any plugins with the same names from the package manager. >> Also, if you ever allow your package manager to apply a software >> "update" to your gstreamer packages, it will happily overwrite your >> work. To prevent that, you may want to place a "hold" on these >> packages to indicate they should not be upgraded. Again, a question >> for your distribution support. For this option you'll also need the >> development packages of the plugins' many dependencies. Depending on >> how thorough your distribution is, this method could take a lot of >> work. >> >> GStreamer is just a set of files -- shared libraries, mostly. Treat it >> as such. A "clean install", intuitively, consists of deleting all the >> current files and installing new ones to replace them. If you use your >> package manager, it does that automatically and lets you think of it >> in terms of "installed" and "uninstalled" (an unnecessary abstraction >> for this case). If you compile from source, you're just overwriting >> the files that the package manager claims to have "installed". >> >> -Sean >> >> On Sat, Sep 5, 2009 at 5:06 PM, Yiliang Bao wrote: >> >>> Hi, >>> >>> I compiled and installed the latest gst-rtsp-server, and updated >>> gstreamer >>> itself based on the error messages from gst-rtsp-server installation. >>> However, after these steps, I could no longer find some plugins like >>> mad, >>> xvimagesink, v4l2src, etc. >>> >>> I have tried re-installing the latest versions of all plugin packages. I >>> have also tried to remove, then re-install all plugins from Synaptic >>> Package >>> Manager. Got some plugin, like v4l2src, back, but mad, xvimagesink are >>> still >>> missing. autovideosink is there, but some pipleine which works is no >>> longer >>> working. For example: >>> >>> gst-launch v4l2src device=/dev/video0 ! autovideosink >>> >>> >>>>>>>>> Error messages: >>>>>>>>> >>> Setting pipeline to PAUSED ... >>> ERROR: Pipeline doesn't want to pause. >>> ERROR: from element >>> /GstPipeline:pipeline0/GstAutoVideoSink:autovideosink0: >>> Could not initialize supporting library. >>> Additional debug info: >>> gstautovideosink.c(373): gst_auto_video_sink_detect (): >>> /GstPipeline:pipeline0/GstAutoVideoSink:autovideosink0: >>> Failed to set target pad >>> Setting pipeline to NULL ... >>> Freeing pipeline ... >>> >>> gst-launch filesrc location=Videos/xyz.avi ! decodebin ! autovideosink >>> >>> >>>>>>>>>>>> Error messages: >>>>>>>>>>>> >>> Setting pipeline to PAUSED ... >>> Pipeline is PREROLLING ... >>> ERROR: from element >>> /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstAviDemux:avidemux0: >>> Internal data stream error. >>> Additional debug info: >>> gstavidemux.c(4443): gst_avi_demux_loop (): >>> /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstAviDemux:avidemux0: >>> streaming stopped, reason not-linked >>> ERROR: pipeline doesn't want to preroll. >>> Setting pipeline to NULL ... >>> Freeing pipeline ... >>> >>> Does anyone know why? Is it possible to re-install a clean version of >>> gstreamer? >>> >>> Yiliang >>> -- >>> View this message in context: >>> http://www.nabble.com/how-to-cleanly-re-install-gstreamer--Help%21%21-tp25252421p25252421.html >>> Sent from the GStreamer-devel mailing list archive at Nabble.com. >>> > > Also, it may be worthwhile to erase your plugin cache and let GStreamer > regenerate it. > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 > 30-Day > trial. Simplify your report design, integration and deployment - and focus > on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- View this message in context: http://www.nabble.com/how-to-cleanly-re-install-gstreamer--Help%21%21-tp25252421p25359622.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From smcnam at gmail.com Wed Sep 9 11:19:38 2009 From: smcnam at gmail.com (Sean McNamara) Date: Wed, 9 Sep 2009 05:19:38 -0400 Subject: [gst-devel] how to cleanly re-install gstreamer? Help!! In-Reply-To: <25359622.post@talk.nabble.com> References: <25252421.post@talk.nabble.com> <74eb1fe20909080840i6606a0aaob90ba78e75a525cd@mail.gmail.com> <4AA68506.2030603@gmail.com> <25359622.post@talk.nabble.com> Message-ID: <74eb1fe20909090219l613a03b1ta960e1812695b9b7@mail.gmail.com> On Wed, Sep 9, 2009 at 2:59 AM, Yiliang Bao wrote: > > Hi Sean, > > Thanks a lot for the detailed explanation. > > I tried both options. Here is what I encountered. Some of information might > be helpful for other people who gets into the same trouble. > > What I did --> > > Originally I installed all the packages using "apt-get ****" from command > line. So they should be official releases. I upgraded gstreamer by compiling > the latest source, it worked fine after resolving some library issues. After > I ran ./configure for the latest base plugin package before I ran make and > install, I found that all the old plugins (almost) were gone. I was able to > resolve most dependencies by installing the development libraries > (libXXX-dev) except for 4 plugins related to header files. After I installed > the base set, I moved on to the good set, and found more problems. I > patiently tried to solve all the library related dependency issues, but was > left with more plugins that could not be built because of header file > related issues. > > After I finished upgrading base and good set. I tried to construct some > pipelines with plugins already installed after my best efforts mentioned > above, but no luck. These pipelines that were working previously no longer > worked. > > 1. Reinstall the old packages from package manager > > After I messed up with gstreamer, I tried to install the old packages using > "apt-get" command, but it said all the packages are the latest, even though > I knew most plugins were already removed. You could just use "aptitude reinstall" (I think even apt-get has a reinstall command). That will put a new copy of the package on top of the old one. Not sure if it purges it first, but in this case it shouldn't matter. > > So this option seems to have some problem. > > 2. Continue compiling from source > > After wasting quite some time, I decided to move ahead with other packages > by ignoring these plugins that I had no idea how to resolve the issues. > Resolving library related dependency issues was rather tedious, but it is > fairly straightforward. I just tried to find the development libraries in > package manager with name similar to that complained in the ./configure > output. > > Finally, I was able to able to install all the packages. Although many > plugins were left un-installed, gstreamer works fine for the functions I > tested, and RTSP server is also working now. > > Thanks again for all the replies! > > Yiliang > > > > L.R.N wrote: >> >> Sean McNamara wrote: >>> Generally it's not a good idea to switch out a different version of >>> the GStreamer core under the feet of installed plugins. When you do >>> that, you should rebuild the plugins. >>> >>> I'm assuming you are on Linux since you use Synaptic Package Manager. >>> >>> You have two options to recover a working gstreamer installation: >>> >>> 1. Reinstall all gstreamer packages from the package manager. You will >>> lose the ability to use gst-rtsp-server and your compiled gstreamer >>> core will be overwritten with what's in the package manager. Figuring >>> out the names of every gstreamer package in order to reinstall them is >>> a question you should ask to your Linux distribution support channel. >>> >>> 2. Compile all the desired gstreamer plugins from source. This will >>> overwrite any plugins with the same names from the package manager. >>> Also, if you ever allow your package manager to apply a software >>> "update" to your gstreamer packages, it will happily overwrite your >>> work. To prevent that, you may want to place a "hold" on these >>> packages to indicate they should not be upgraded. Again, a question >>> for your distribution support. For this option you'll also need the >>> development packages of the plugins' many dependencies. Depending on >>> how thorough your distribution is, this method could take a lot of >>> work. >>> >>> GStreamer is just a set of files -- shared libraries, mostly. Treat it >>> as such. A "clean install", intuitively, consists of deleting all the >>> current files and installing new ones to replace them. If you use your >>> package manager, it does that automatically and lets you think of it >>> in terms of "installed" and "uninstalled" (an unnecessary abstraction >>> for this case). If you compile from source, you're just overwriting >>> the files that the package manager claims to have "installed". >>> >>> -Sean >>> >>> On Sat, Sep 5, 2009 at 5:06 PM, Yiliang Bao wrote: >>> >>>> Hi, >>>> >>>> I compiled and installed the latest gst-rtsp-server, and updated >>>> gstreamer >>>> itself based on the error messages from gst-rtsp-server installation. >>>> However, after these steps, I could no longer find some plugins like >>>> mad, >>>> xvimagesink, v4l2src, etc. >>>> >>>> I have tried re-installing the latest versions of all plugin packages. I >>>> have also tried to remove, then re-install all plugins from Synaptic >>>> Package >>>> Manager. Got some plugin, like v4l2src, back, but mad, xvimagesink are >>>> still >>>> missing. autovideosink is there, but some pipleine which works is no >>>> longer >>>> working. For example: >>>> >>>> gst-launch v4l2src device=/dev/video0 ! autovideosink >>>> >>>> >>>>>>>>>> Error messages: >>>>>>>>>> >>>> Setting pipeline to PAUSED ... >>>> ERROR: Pipeline doesn't want to pause. >>>> ERROR: from element >>>> /GstPipeline:pipeline0/GstAutoVideoSink:autovideosink0: >>>> Could not initialize supporting library. >>>> Additional debug info: >>>> gstautovideosink.c(373): gst_auto_video_sink_detect (): >>>> /GstPipeline:pipeline0/GstAutoVideoSink:autovideosink0: >>>> Failed to set target pad >>>> Setting pipeline to NULL ... >>>> Freeing pipeline ... >>>> >>>> gst-launch filesrc location=Videos/xyz.avi ! decodebin ! autovideosink >>>> >>>> >>>>>>>>>>>>> Error messages: >>>>>>>>>>>>> >>>> Setting pipeline to PAUSED ... >>>> Pipeline is PREROLLING ... >>>> ERROR: from element >>>> /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstAviDemux:avidemux0: >>>> Internal data stream error. >>>> Additional debug info: >>>> gstavidemux.c(4443): gst_avi_demux_loop (): >>>> /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstAviDemux:avidemux0: >>>> streaming stopped, reason not-linked >>>> ERROR: pipeline doesn't want to preroll. >>>> Setting pipeline to NULL ... >>>> Freeing pipeline ... >>>> >>>> Does anyone know why? Is it possible to re-install a clean version of >>>> gstreamer? >>>> >>>> Yiliang >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/how-to-cleanly-re-install-gstreamer--Help%21%21-tp25252421p25252421.html >>>> Sent from the GStreamer-devel mailing list archive at Nabble.com. >>>> >> >> Also, it may be worthwhile to erase your plugin cache and let GStreamer >> regenerate it. >> >> ------------------------------------------------------------------------------ >> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 >> 30-Day >> trial. Simplify your report design, integration and deployment - and focus >> on >> what you do best, core application coding. Discover what's new with >> Crystal Reports now. ?http://p.sf.net/sfu/bobj-july >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > -- > View this message in context: http://www.nabble.com/how-to-cleanly-re-install-gstreamer--Help%21%21-tp25252421p25359622.html > Sent from the GStreamer-devel mailing list archive at Nabble.com. > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. ?http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From vitaly.v.ch at gmail.com Wed Sep 9 14:00:54 2009 From: vitaly.v.ch at gmail.com (Vitaly V. Ch) Date: Wed, 9 Sep 2009 15:00:54 +0300 Subject: [gst-devel] partially upgrating Gst-plugins Message-ID: <6efe08af0909090500u3703bec5o15f8a7dad92769cd@mail.gmail.com> Hi all!-) I can upgrade only gst-plugins-bad without upgrading gstreamer or gst-plugin-base? \\wbr Vitaly Chernookiy From bilboed at gmail.com Wed Sep 9 14:54:25 2009 From: bilboed at gmail.com (Edward Hervey) Date: Wed, 09 Sep 2009 14:54:25 +0200 Subject: [gst-devel] partially upgrating Gst-plugins In-Reply-To: <6efe08af0909090500u3703bec5o15f8a7dad92769cd@mail.gmail.com> References: <6efe08af0909090500u3703bec5o15f8a7dad92769cd@mail.gmail.com> Message-ID: <1252500865.16797.93.camel@localhost> On Wed, 2009-09-09 at 15:00 +0300, Vitaly V. Ch wrote: > Hi all!-) > > I can upgrade only gst-plugins-bad without upgrading gstreamer or > gst-plugin-base? If the new gst-plugins-bad doesn't require a more recent version of gstreamer or gst-p-b then yes.... else no. There's no way around it, recent versions of gst-plugins-bad will make usage of features in newer gstreamer/gst-p-b. Edward > > \\wbr Vitaly Chernookiy > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From rakesh2.sharma at aricent.com Wed Sep 9 14:50:05 2009 From: rakesh2.sharma at aricent.com (rakesh sharma) Date: Wed, 9 Sep 2009 05:50:05 -0700 (PDT) Subject: [gst-devel] CPU Load bcoz of Gstreamer plugin In-Reply-To: <25346008.post@talk.nabble.com> References: <25340716.post@talk.nabble.com> <1252397608.2148.7.camel@putamadre> <25343663.post@talk.nabble.com> <1252408058.16797.24.camel@localhost> <25346008.post@talk.nabble.com> Message-ID: <25364310.post@talk.nabble.com> I am still not able to analyse where its eating full cpu .... If anybody is having any idea please share with me.... rakesh sharma wrote: > > On device there is no debug mode supported. > Thanks for that googling thing i'll search for those tools. > Any other idea... > > > Edward Hervey wrote: >> >> On Tue, 2009-09-08 at 03:44 -0700, rakesh sharma wrote: >>> We dont have those tools callgrind/oprofile on the target platform. >>> I have written the plugin in lcml(socket node), so code should take >>> maximum >>> cpu on dsp side not arm side, but its tacking full cpu on arm side . I >>> am >>> not able to measure where its eating cpu. >>> If you people have those tools for arm you can share it with me so that >>> i'll >>> check and let you know about that.please reply soon its urgent. >> >> 1. I just can not believe your employer doesn't have profiling tools for >> ARM. (Googling for "arm profiling linux" returns heaps of >> results/tools). >> >> 2. If you'd carried on reading my mail you would have read "or analyze >> your logs to see where it's taking all that time". By logs, I mean >> GST_DEBUG logs. >> >> Edward >> >>> >>> thanks for reply... >>> >>> >>> Edward Hervey wrote: >>> > >>> > Sorry to ask the obvious question... >>> > >>> > ... but have you profiled your pipeline and/or encoder to see *WHERE* >>> > it's taking 99% cpu ? Use a profiler like callgrind/oprofile (if >>> > available on your platform) or analyze your logs to see where it's >>> > taking all that time. >>> > >>> > Until you've analyzed where your problem is... we won't be able to >>> > help you. >>> > >>> > On Mon, 2009-09-07 at 23:47 -0700, rakesh sharma wrote: >>> >> I have written mp4sp encoder and its tacking 99% cpu load for all >>> >> resolutions. >>> >> Can anyone suggest solution for this ? >>> >> Is that i have to release cpu using some gstreamer api ? >>> >> Please help me its urgent..... >>> > >>> > >>> > >>> ------------------------------------------------------------------------------ >>> > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 >>> > 30-Day >>> > trial. Simplify your report design, integration and deployment - and >>> focus >>> > on >>> > what you do best, core application coding. Discover what's new with >>> > Crystal Reports now. http://p.sf.net/sfu/bobj-july >>> > _______________________________________________ >>> > gstreamer-devel mailing list >>> > gstreamer-devel at lists.sourceforge.net >>> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> > >>> > >>> >> >> >> ------------------------------------------------------------------------------ >> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 >> 30-Day >> trial. Simplify your report design, integration and deployment - and >> focus on >> what you do best, core application coding. Discover what's new with >> Crystal Reports now. http://p.sf.net/sfu/bobj-july >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > -- View this message in context: http://www.nabble.com/CPU-Load-bcoz-of-Gstreamer-plugin-tp25340716p25364310.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From vitaly.v.ch at gmail.com Wed Sep 9 14:56:49 2009 From: vitaly.v.ch at gmail.com (Vitaly V. Ch) Date: Wed, 9 Sep 2009 15:56:49 +0300 Subject: [gst-devel] partially upgrating Gst-plugins In-Reply-To: <1252500865.16797.93.camel@localhost> References: <6efe08af0909090500u3703bec5o15f8a7dad92769cd@mail.gmail.com> <1252500865.16797.93.camel@localhost> Message-ID: <6efe08af0909090556n5c6229a2p150bb402459f23c@mail.gmail.com> On Wed, Sep 9, 2009 at 3:54 PM, Edward Hervey wrote: > On Wed, 2009-09-09 at 15:00 +0300, Vitaly V. Ch wrote: >> Hi all!-) >> >> I can upgrade only gst-plugins-bad without upgrading gstreamer or >> gst-plugin-base? > > ?If the new gst-plugins-bad doesn't require a more recent version of > gstreamer or gst-p-b then yes.... else no. Where I can found requirements for gst-plugins-bad? > > ?There's no way around it, recent versions of gst-plugins-bad will make > usage of features in newer gstreamer/gst-p-b. > > ? ?Edward > >> >> \\wbr Vitaly Chernookiy >> >> ------------------------------------------------------------------------------ >> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day >> trial. Simplify your report design, integration and deployment - and focus on >> what you do best, core application coding. Discover what's new with >> Crystal Reports now. ?http://p.sf.net/sfu/bobj-july >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. ?http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From bilboed at gmail.com Wed Sep 9 17:12:57 2009 From: bilboed at gmail.com (Edward Hervey) Date: Wed, 09 Sep 2009 17:12:57 +0200 Subject: [gst-devel] partially upgrating Gst-plugins In-Reply-To: <6efe08af0909090556n5c6229a2p150bb402459f23c@mail.gmail.com> References: <6efe08af0909090500u3703bec5o15f8a7dad92769cd@mail.gmail.com> <1252500865.16797.93.camel@localhost> <6efe08af0909090556n5c6229a2p150bb402459f23c@mail.gmail.com> Message-ID: <1252509177.16797.116.camel@localhost> On Wed, 2009-09-09 at 15:56 +0300, Vitaly V. Ch wrote: > On Wed, Sep 9, 2009 at 3:54 PM, Edward Hervey wrote: > > On Wed, 2009-09-09 at 15:00 +0300, Vitaly V. Ch wrote: > >> Hi all!-) > >> > >> I can upgrade only gst-plugins-bad without upgrading gstreamer or > >> gst-plugin-base? > > > > If the new gst-plugins-bad doesn't require a more recent version of > > gstreamer or gst-p-b then yes.... else no. > > Where I can found requirements for gst-plugins-bad? in configure.ac Edward > > > > > There's no way around it, recent versions of gst-plugins-bad will make > > usage of features in newer gstreamer/gst-p-b. > > > > Edward > > > >> > >> \\wbr Vitaly Chernookiy > >> > >> ------------------------------------------------------------------------------ > >> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > >> trial. Simplify your report design, integration and deployment - and focus on > >> what you do best, core application coding. Discover what's new with > >> Crystal Reports now. http://p.sf.net/sfu/bobj-july > >> _______________________________________________ > >> gstreamer-devel mailing list > >> gstreamer-devel at lists.sourceforge.net > >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > ------------------------------------------------------------------------------ > > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > > trial. Simplify your report design, integration and deployment - and focus on > > what you do best, core application coding. Discover what's new with > > Crystal Reports now. http://p.sf.net/sfu/bobj-july > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From bilboed at gmail.com Wed Sep 9 17:12:37 2009 From: bilboed at gmail.com (Edward Hervey) Date: Wed, 09 Sep 2009 17:12:37 +0200 Subject: [gst-devel] CPU Load bcoz of Gstreamer plugin In-Reply-To: <25364310.post@talk.nabble.com> References: <25340716.post@talk.nabble.com> <1252397608.2148.7.camel@putamadre> <25343663.post@talk.nabble.com> <1252408058.16797.24.camel@localhost> <25346008.post@talk.nabble.com> <25364310.post@talk.nabble.com> Message-ID: <1252509157.16797.115.camel@localhost> On Wed, 2009-09-09 at 05:50 -0700, rakesh sharma wrote: > I am still not able to analyse where its eating full cpu .... You will have to persevere then. We can't guess where you problem is. Edward > If anybody is having any idea please share with me.... > > > rakesh sharma wrote: > > > > On device there is no debug mode supported. > > Thanks for that googling thing i'll search for those tools. > > Any other idea... > > > > > > Edward Hervey wrote: > >> > >> On Tue, 2009-09-08 at 03:44 -0700, rakesh sharma wrote: > >>> We dont have those tools callgrind/oprofile on the target platform. > >>> I have written the plugin in lcml(socket node), so code should take > >>> maximum > >>> cpu on dsp side not arm side, but its tacking full cpu on arm side . I > >>> am > >>> not able to measure where its eating cpu. > >>> If you people have those tools for arm you can share it with me so that > >>> i'll > >>> check and let you know about that.please reply soon its urgent. > >> > >> 1. I just can not believe your employer doesn't have profiling tools for > >> ARM. (Googling for "arm profiling linux" returns heaps of > >> results/tools). > >> > >> 2. If you'd carried on reading my mail you would have read "or analyze > >> your logs to see where it's taking all that time". By logs, I mean > >> GST_DEBUG logs. > >> > >> Edward > >> > >>> > >>> thanks for reply... > >>> > >>> > >>> Edward Hervey wrote: > >>> > > >>> > Sorry to ask the obvious question... > >>> > > >>> > ... but have you profiled your pipeline and/or encoder to see *WHERE* > >>> > it's taking 99% cpu ? Use a profiler like callgrind/oprofile (if > >>> > available on your platform) or analyze your logs to see where it's > >>> > taking all that time. > >>> > > >>> > Until you've analyzed where your problem is... we won't be able to > >>> > help you. > >>> > > >>> > On Mon, 2009-09-07 at 23:47 -0700, rakesh sharma wrote: > >>> >> I have written mp4sp encoder and its tacking 99% cpu load for all > >>> >> resolutions. > >>> >> Can anyone suggest solution for this ? > >>> >> Is that i have to release cpu using some gstreamer api ? > >>> >> Please help me its urgent..... > >>> > > >>> > > >>> > > >>> ------------------------------------------------------------------------------ > >>> > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 > >>> > 30-Day > >>> > trial. Simplify your report design, integration and deployment - and > >>> focus > >>> > on > >>> > what you do best, core application coding. Discover what's new with > >>> > Crystal Reports now. http://p.sf.net/sfu/bobj-july > >>> > _______________________________________________ > >>> > gstreamer-devel mailing list > >>> > gstreamer-devel at lists.sourceforge.net > >>> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > >>> > > >>> > > >>> > >> > >> > >> ------------------------------------------------------------------------------ > >> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 > >> 30-Day > >> trial. Simplify your report design, integration and deployment - and > >> focus on > >> what you do best, core application coding. Discover what's new with > >> Crystal Reports now. http://p.sf.net/sfu/bobj-july > >> _______________________________________________ > >> gstreamer-devel mailing list > >> gstreamer-devel at lists.sourceforge.net > >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > >> > >> > > > > > From vitaly.v.ch at gmail.com Wed Sep 9 17:16:16 2009 From: vitaly.v.ch at gmail.com (Vitaly V. Ch) Date: Wed, 9 Sep 2009 18:16:16 +0300 Subject: [gst-devel] partially upgrating Gst-plugins In-Reply-To: <1252509177.16797.116.camel@localhost> References: <6efe08af0909090500u3703bec5o15f8a7dad92769cd@mail.gmail.com> <1252500865.16797.93.camel@localhost> <6efe08af0909090556n5c6229a2p150bb402459f23c@mail.gmail.com> <1252509177.16797.116.camel@localhost> Message-ID: <6efe08af0909090816m634c06dewa491727b97ec06b0@mail.gmail.com> Thanks, \\wbr Vitaly Chernookiy On Wed, Sep 9, 2009 at 6:12 PM, Edward Hervey wrote: > On Wed, 2009-09-09 at 15:56 +0300, Vitaly V. Ch wrote: >> On Wed, Sep 9, 2009 at 3:54 PM, Edward Hervey wrote: >> > On Wed, 2009-09-09 at 15:00 +0300, Vitaly V. Ch wrote: >> >> Hi all!-) >> >> >> >> I can upgrade only gst-plugins-bad without upgrading gstreamer or >> >> gst-plugin-base? >> > >> > ?If the new gst-plugins-bad doesn't require a more recent version of >> > gstreamer or gst-p-b then yes.... else no. >> >> Where I can found requirements for gst-plugins-bad? > > ? in configure.ac > > ? ? Edward > >> >> > >> > ?There's no way around it, recent versions of gst-plugins-bad will make >> > usage of features in newer gstreamer/gst-p-b. >> > >> > ? ?Edward >> > >> >> >> >> \\wbr Vitaly Chernookiy >> >> >> >> ------------------------------------------------------------------------------ >> >> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day >> >> trial. Simplify your report design, integration and deployment - and focus on >> >> what you do best, core application coding. Discover what's new with >> >> Crystal Reports now. ?http://p.sf.net/sfu/bobj-july >> >> _______________________________________________ >> >> gstreamer-devel mailing list >> >> gstreamer-devel at lists.sourceforge.net >> >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > >> > >> > ------------------------------------------------------------------------------ >> > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day >> > trial. Simplify your report design, integration and deployment - and focus on >> > what you do best, core application coding. Discover what's new with >> > Crystal Reports now. ?http://p.sf.net/sfu/bobj-july >> > _______________________________________________ >> > gstreamer-devel mailing list >> > gstreamer-devel at lists.sourceforge.net >> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > >> >> ------------------------------------------------------------------------------ >> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day >> trial. Simplify your report design, integration and deployment - and focus on >> what you do best, core application coding. Discover what's new with >> Crystal Reports now. ?http://p.sf.net/sfu/bobj-july >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. ?http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From virajk at gmail.com Wed Sep 9 19:33:46 2009 From: virajk at gmail.com (Viraj Karandikar) Date: Wed, 9 Sep 2009 23:03:46 +0530 Subject: [gst-devel] AEC plugin in Gstreamer In-Reply-To: <1251656835.14075.5.camel@TesterTop3.tester.ca> References: <25129703.post@talk.nabble.com> <4A9AC1F9.1010408@hora-obscura.de> <1251656835.14075.5.camel@TesterTop3.tester.ca> Message-ID: Hi, We are developing AEC plugin for our proprietary AEC implementation. We are having 2 sink pads (one for near end and one for far end inputs) and 1 source pad (for AEC output). Audio capture and playback can happen with any of the available plugins. But you have to implement a logic to make sure that the input data to AEC algo is in correct sync. Also is it required to have as minimum delay as possible in capture and playback path to have short tail length. Regards, Viraj 2009/8/30 Olivier Cr?te > On Sun, 2009-08-30 at 21:16 +0300, Stefan Kost wrote: > > rmkart schrieb: > > > Hi, > > > Is there any AEC (Achoustic echo cancellation) Gstreamer plugin > available. > > > For this I need to get the input from alsasink and send it to Alsasrc, > the > > > Alsa component does a buffering within itsself. If I try to apply AEC > on > > > these buffers then the quality wont be good as the logic is pplies on > > > buffers which might not be in syc. Can anyone siggest me any > > > ideas/suggestion for this. > > > Thanks, > > > RK > > > > There is none I am aware of. If you consider start making one, keep us > up-to-date. > > I started writing one based on libspeexdsp, but I never got it to work > properly. The code is at: > > > http://git.collabora.co.uk/?p=user/tester/gst-plugins-farsight-tester.git;a=tree;f=ext/speexdsp;hb=speexdsp > > Also, my understanding is that the AEC algorithm in speexdsp will only > work if the src and the sink are on the same sound card. So it won't > work with stuff like USB or Bluetooth speakers or with the microphone on > a Webcam. So I kind of gave up on it. > > -- > Olivier Cr?te > olivier.crete at collabora.co.uk > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus > on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- - Viraj Reality is merely an illusion, albeit a very persistent one. -------------- next part -------------- An HTML attachment was scrubbed... URL: From olivier.crete at collabora.co.uk Wed Sep 9 19:44:58 2009 From: olivier.crete at collabora.co.uk (Olivier =?ISO-8859-1?Q?Cr=EAte?=) Date: Wed, 09 Sep 2009 13:44:58 -0400 Subject: [gst-devel] AEC plugin in Gstreamer In-Reply-To: References: <25129703.post@talk.nabble.com> <4A9AC1F9.1010408@hora-obscura.de> <1251656835.14075.5.camel@TesterTop3.tester.ca> Message-ID: <1252518298.30756.7.camel@TesterTop3.tester.ca> On Wed, 2009-09-09 at 23:03 +0530, Viraj Karandikar wrote: > Hi, > We are developing AEC plugin for our proprietary AEC implementation. > We are having 2 sink pads (one for near end and one for far end > inputs) and 1 source pad (for AEC output). > Audio capture and playback can happen with any of the available > plugins. > But you have to implement a logic to make sure that the input data to > AEC algo is in correct sync. > Also is it required to have as minimum delay as possible in capture > and playback path to have short tail length. Yes, thats exactly what I was trying to do with the speexdsp based one. But for some reason it never worked properly. So either I'm almost there and there is only some tiny bugs.. Or I'm entirely on the wrong path.. Either way, you have to work it out.. Olivier > 2009/8/30 Olivier Cr?te > On Sun, 2009-08-30 at 21:16 +0300, Stefan Kost wrote: > > rmkart schrieb: > > > Hi, > > > Is there any AEC (Achoustic echo cancellation) Gstreamer > plugin available. > > > For this I need to get the input from alsasink and send it > to Alsasrc, the > > > Alsa component does a buffering within itsself. If I try > to apply AEC on > > > these buffers then the quality wont be good as the logic > is pplies on > > > buffers which might not be in syc. Can anyone siggest me > any > > > ideas/suggestion for this. > > > Thanks, > > > RK > > > > There is none I am aware of. If you consider start making > one, keep us up-to-date. > > > I started writing one based on libspeexdsp, but I never got it > to work > properly. The code is at: > > http://git.collabora.co.uk/?p=user/tester/gst-plugins-farsight-tester.git;a=tree;f=ext/speexdsp;hb=speexdsp > > Also, my understanding is that the AEC algorithm in speexdsp > will only > work if the src and the sink are on the same sound card. So it > won't > work with stuff like USB or Bluetooth speakers or with the > microphone on > a Webcam. So I kind of gave up on it. > > -- > Olivier Cr?te > olivier.crete at collabora.co..uk > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal > Reports 2008 30-Day > trial. Simplify your report design, integration and deployment > - and focus on > what you do best, core application coding. Discover what's new > with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > -- > - Viraj > Reality is merely an illusion, albeit a very persistent one. > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -- Olivier Cr?te olivier.crete at collabora.co.uk -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: This is a digitally signed message part URL: From havard.graff at tandberg.com Wed Sep 9 20:00:01 2009 From: havard.graff at tandberg.com (=?iso-8859-1?Q?H=E5vard_Graff?=) Date: Wed, 9 Sep 2009 20:00:01 +0200 Subject: [gst-devel] AEC plugin in Gstreamer In-Reply-To: <1252518298.30756.7.camel@TesterTop3.tester.ca> References: <25129703.post@talk.nabble.com> <4A9AC1F9.1010408@hora-obscura.de><1251656835.14075.5.camel@TesterTop3.tester.ca> <1252518298.30756.7.camel@TesterTop3.tester.ca> Message-ID: <9F6ACAE02B6DD040A1E259977622CFDB06416267@oslexcp1.eu.tandberg.int> Some tips: 1. Don't try to use existing audio src and sinks. You are dependant on being able to capture and playout in one operation (read: thread), not two independent ones. 2. Depending on the sensitivity of your algorithm, there will often be different clocks (read: crystals) on your capture and playout devices. Even though it says it gives you 48000 samples per second, it is usually lying... Better accuracy = better AEC. 3. Try to estimate the round-trip of your sound from sink to src as close as you can get, and feed this info to the AEC. Good luck! Let us know if you get somewhere. H?vard -----Original Message----- From: Olivier Cr?te [mailto:olivier.crete at collabora.co.uk] Sent: 9. september 2009 19:45 To: gstreamer-devel at lists.sourceforge.net Subject: Re: [gst-devel] AEC plugin in Gstreamer On Wed, 2009-09-09 at 23:03 +0530, Viraj Karandikar wrote: > Hi, > We are developing AEC plugin for our proprietary AEC implementation. > We are having 2 sink pads (one for near end and one for far end > inputs) and 1 source pad (for AEC output). > Audio capture and playback can happen with any of the available > plugins. > But you have to implement a logic to make sure that the input data to > AEC algo is in correct sync. > Also is it required to have as minimum delay as possible in capture > and playback path to have short tail length. Yes, thats exactly what I was trying to do with the speexdsp based one. But for some reason it never worked properly. So either I'm almost there and there is only some tiny bugs.. Or I'm entirely on the wrong path.. Either way, you have to work it out.. Olivier > 2009/8/30 Olivier Cr?te > On Sun, 2009-08-30 at 21:16 +0300, Stefan Kost wrote: > > rmkart schrieb: > > > Hi, > > > Is there any AEC (Achoustic echo cancellation) Gstreamer > plugin available. > > > For this I need to get the input from alsasink and send it > to Alsasrc, the > > > Alsa component does a buffering within itsself. If I try > to apply AEC on > > > these buffers then the quality wont be good as the logic > is pplies on > > > buffers which might not be in syc. Can anyone siggest me > any > > > ideas/suggestion for this. > > > Thanks, > > > RK > > > > There is none I am aware of. If you consider start making > one, keep us up-to-date. > > > I started writing one based on libspeexdsp, but I never got it > to work > properly. The code is at: > > http://git.collabora.co.uk/?p=user/tester/gst-plugins-farsight-tester.git;a=tree;f=ext/speexdsp;hb=speexdsp > > Also, my understanding is that the AEC algorithm in speexdsp > will only > work if the src and the sink are on the same sound card. So it > won't > work with stuff like USB or Bluetooth speakers or with the > microphone on > a Webcam. So I kind of gave up on it. > > -- > Olivier Cr?te > olivier.crete at collabora.co..uk > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal > Reports 2008 30-Day > trial. Simplify your report design, integration and deployment > - and focus on > what you do best, core application coding. Discover what's new > with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > -- > - Viraj > Reality is merely an illusion, albeit a very persistent one. > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -- Olivier Cr?te olivier.crete at collabora.co.uk From vitaly.v.ch at gmail.com Wed Sep 9 20:47:00 2009 From: vitaly.v.ch at gmail.com (Vitaly V. Ch) Date: Wed, 9 Sep 2009 21:47:00 +0300 Subject: [gst-devel] ERROR: from element /pipeline0/timidity0: Could not decode stream. In-Reply-To: <6efe08af0909080836k2a2439bbr888a669cc0ea9584@mail.gmail.com> References: <6efe08af0909060413u42207b0evff87f2a512e010f1@mail.gmail.com> <4AA40A53.4050409@hora-obscura.de> <6efe08af0909062350w31705eb7n1f1e58d4a27f6169@mail.gmail.com> <4AA677E2.9010308@hora-obscura.de> <6efe08af0909080836k2a2439bbr888a669cc0ea9584@mail.gmail.com> Message-ID: <6efe08af0909091147g43644e22jed4014aaf3e4dac4@mail.gmail.com> I report this bug as https://bugzilla.gnome.org/show_bug.cgi?id=594680 \\wbr Vitaly Chernookiy On Tue, Sep 8, 2009 at 6:36 PM, Vitaly V. Ch wrote: > Ok, I will do it tomorrow. > > On Tue, Sep 8, 2009 at 6:27 PM, Stefan Kost wrote: >> hi, >> Vitaly V. Ch schrieb: >>> This midi file played successfully via aplaymidi but on my system >>> gstreamer can't play midi files at all. >>> >>> I'm beginner in Gstreamer and in this case need guide. >>> >> well file a bug and attach the file. Then its easier to help. >> >> Stefan >> >>> \\wbr Vitaly >>> >>> On Sun, Sep 6, 2009 at 10:15 PM, Stefan Kost wrote: >>> >>>> Vitaly V. Ch schrieb: >>>> >>>>> I want to play midi via gstreamer but got next trouble: >>>>> >>>>> # LANG= gst-launch filesrc location=Krysha_doma_tvoego.mid ! timidity ! alsasink >>>>> Setting pipeline to PAUSED ... >>>>> Pipeline is PREROLLING ... >>>>> ERROR: from element /pipeline0/timidity0: Could not decode stream. >>>>> Additional debug info: >>>>> gsttimidity.c(641): gst_timidity_loop (): /pipeline0/timidity0: >>>>> Unable to parse midi >>>>> ERROR: pipeline doesn't want to preroll. >>>>> Setting pipeline to NULL ... >>>>> FREEING pipeline ... >>>>> # >>>>> >>>>> >>>> can you play that with other players. Please double check that its a midi-file >>>> (what does "file Krysha_doma_tvoego.mid" reports). If it only fails with >>>> gstreamer, please file a bug and attach the file. >>>> >>>> Stefan >>>> >>>> ------------------------------------------------------------------------------ >>>> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day >>>> trial. Simplify your report design, integration and deployment - and focus on >>>> what you do best, core application coding. Discover what's new with >>>> Crystal Reports now. ?http://p.sf.net/sfu/bobj-july >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> gstreamer-devel at lists.sourceforge.net >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>>> >>> >>> ------------------------------------------------------------------------------ >>> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day >>> trial. Simplify your report design, integration and deployment - and focus on >>> what you do best, core application coding. Discover what's new with >>> Crystal Reports now. ?http://p.sf.net/sfu/bobj-july >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >> >> >> ------------------------------------------------------------------------------ >> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day >> trial. Simplify your report design, integration and deployment - and focus on >> what you do best, core application coding. Discover what's new with >> Crystal Reports now. ?http://p.sf.net/sfu/bobj-july >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > From rob at ti.com Thu Sep 10 02:36:43 2009 From: rob at ti.com (Rob Clark) Date: Wed, 9 Sep 2009 19:36:43 -0500 Subject: [gst-devel] proposal: support for row-stride in gstreamer In-Reply-To: References: <00975935-AF31-49B4-A61A-B93614ED3963@ti.com> <1248251000.25570.1.camel@fancy.localdomain> <8CF1F328-F9FA-46B3-9C03-1936A261C72F@ti.com> <1249379361.2756.45.camel@fancy.localdomain> <7c34ac520908102237g514c9fb7h3f4a61fe18458712@mail.gmail.com> Message-ID: <5A29FACE-C96A-4F7C-B892-92324A3BCE96@ti.com> Just to update.. I have pipelines with rowstride working fine. The approach is relatively straightforward and not too revolutionary. Just normal caps negotiation, which I guess should not be too surprising. I did add some utility functions and macros in libgstvideo to help with the caps building/parsing. And I did add a patch on gst_pad_fixate_caps() (see below). Other than gst_pad_fixate_caps, the changes are all in gst-plugins-base. You can find my git tree: git://github.com/robclark/gst-plugins-base.git http://github.com/robclark/gst-plugins-base/commits/master (I agree with earlier comments that stride-transformation should be supported in ffmpegcolorspace.. but this part I've not had a chance to implement yet.) ------------------------------------------ Essentially what is required is for a video playback pipeline: 1) srcpad on video decoders, sinkpad on video sinks, etc, should add x- raw-yuv-strided to their template caps: static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS (GST_VIDEO_CAPS_YUV_STRIDED ( "{ I420, YUY2, UYVY }", "[ 0, max ]")) ); 2) elements like the video decoders should implement a _get_caps() function for the srcpad which returns two equivalent structures, video/ x-raw-yuv and video/x-raw-yuv-strided (the later with a rowstride field). It might be worthwhile to add a helper function in libgstvideo for this. 3) And in the _set_caps() function for video decoder srcpad should do something like: if (gst_video_format_parse_caps_strided (caps, &format, &width, &height, &rowstride)) { ... configure decoder width/height/fourcc ... if (rowstride) { ... configure decoder rowstride ... } } and of course the video decoder should pad_alloc() their buffers to give the video sink element an opportunity to dictate the rowstride that it would prefer to use, if any. 4) To have *some* caps when allocating the first buffer, I used something like: new_caps = gst_caps_intersect (gst_pad_get_caps (srcpad), gst_pad_peer_get_caps (srcpad)); if (!gst_caps_is_fixed (new_caps)) { gst_caps_do_simplify (new_caps); gst_pad_fixate_caps (srcpad, new_caps); } gst_pad_set_caps (srcpad, new_caps); Now here is where I ran into a small issue that I think is best fixed in gst_pad_fixate_caps(). If the caps consists of multiple structs (such as strided and regular non-strided caps), gst_pad_fixate_caps() will only fixate the individual structs, and not choose a single struct to make the caps fixed. I made a small change to remove all but the first struct: ------------------------------------------ gst_pad_fixate_caps (GstPad * pad, GstCaps * caps) { GstPadFixateCapsFunction fixatefunc; - guint n, len; + guint len; g_return_if_fail (GST_IS_PAD (pad)); g_return_if_fail (caps != NULL); @@ -2359,11 +2359,15 @@ gst_pad_fixate_caps (GstPad * pad, GstCaps * caps) /* default fixation */ len = gst_caps_get_size (caps); - for (n = 0; n < len; n++) { - GstStructure *s = gst_caps_get_structure (caps, n); + if (len > 0) { + GstStructure *s = gst_caps_get_structure (caps, 0); gst_structure_foreach (s, gst_pad_default_fixate, s); } + + while (len > 1) { + gst_caps_remove_structure (caps, --len); + } } ------------------------------------------ I'm curious if this is the correct solution, or if there is some use- case for a function which fixates the individual structs but does not make the caps fixed? BR, -R On Aug 11, 2009, at 2:03 PM, Clark, Rob wrote: > Hi Jun, > > I would like to avoid subclassing GstBuffer for this.. since a lot of > videosink (and other) elements already subclass GstBuffer for various > purposes. Having to both subclass GstBuffer and GstVideoBuffer would > make a mess. > > If there was some case where the stride could be changing from frame > to frame, the buffer metadata proposal would solve this: > > http://cgit.freedesktop.org/gstreamer/gstreamer/tree/docs/design/draft-buffer2.txt > > but, at least in the cases I can think of, the stride would not be > changing from frame to frame. > > I'm currently working on implementing support for it in gst-openmax > video decoder elements.. which at least should serve as a good > reference to the changes to make in other video decoder/encoder > elements. I think it won't be so complicated, although the decoder > element would need to support re-negotiating the caps when it > pad_alloc's a buffer from the video sink element. But this shouldn't > be a big deal. > > ---- > > as far as stridetransform.. I wouldn't expect it to be part of a > "normal" pipeline. I'm using it for now for testing. If the > consensus is that normal pipelines might need some stride > transformation, I think it should be combined with colorspace > conversion, to avoid multiple passes over the decoded video frame. > (But I've not had a chance to implement this yet.) > > BR, > -R > > > On Aug 11, 2009, at 12:37 AM, Jim Nie wrote: > >> I agree to use x-raw-yuv-stride, x-raw-rgb-stride to mark the >> buffer and pad capability. However, stridetransform element may >> impact >> the other elements and application usage. It will be big effort to >> adopt it. >> >> I have a propose that may not be mature neither. Comments are >> welcome. >> Elements that support stride, or sub region of video buffers will >> support both x-raw-yuv-stride and x-raw-yuv. If successfully >> negotiated with downstream element, its src_pad will use >> gst_video_buffer that derived from gst_buffer. gst_video_buffer >> contains the offset/stride information. >> >> If element fails to negotiate with downstream element with >> x-raw-yuv-stride, it will try x-raw-yuv as current model. Memory is >> copied from subregion to new malloced memory and push to downstream. >> >> In this way, we can keep compatible with current framework. All >> effort introduced by supporting subregion/stride is limited to the >> elements that want to support it. >> >> Jun >> >> 2009/8/4, Rob Clark : >>> >>> On Aug 4, 2009, at 4:49 AM, Jan Schmidt wrote: >>> >>>> Hi, >>>> >>>> On Thu, 2009-07-30 at 11:45 -0500, Rob Clark wrote: >>>>> so, my next steps (at least my current thinking about next steps) >>>>> are >>>>> to start adding some rowstride related stuff in gst-plugins-base: >>>>> >>>>> >>>>> gst-libs/gst/video/video.c: >>>>> 1) add gst_video_format_get_size_strided() >>>>> 2) change gst_video_format_parse_caps() to understand strided >>>>> caps, >>>>> and add a gst_video_format_parse_caps_strided() function (it >>>>> seems >>>>> there are enough places already using >>>>> gst_video_format_parse_caps() >>>>> that I probably don't want to change the signature of this >>>>> function) >>>>> >>>>> and then add a GstStrideTransform element under gst/stride >>>> >>>> I haven't put enough thought into designing a stride system to say >>>> anything for certain. Particularly, I'm not sure whether we'll be >>>> able >>>> to safely and successfully integrate it into the 0.10 series >>>> without >>>> thinking about it harder. >>>> >>>> I am pretty sure, however, that adding a separate stride adjust >>>> element >>>> is the wrong way to go. None of the existing pipelines will include >>>> it, >>>> so it would no be useful without changes to the applications/ >>>> playbin >>>> etc. >>>> >>>> My hunch is that it will be better to add a new 'video/x-raw-yuv- >>>> full' >>>> format and add stride as a new parameter, and adjust >>>> ffmpegcolorspace to >>>> support conversions to/from the unstrided/strided formats. >>> >>> >>> Hmm.. my idea for stridetransform was mainly just an element for >>> testing and debugging with manually constructed gst-launch >>> pipelines. >>> We need some way to run codecs both with and without rowstide and >>> verifying the output. For example: >>> >>> >>> ... ! decoder ! video/x-raw- >>> yuv,format=(fourcc)YUY2,width=320,height=240,framerate=30/1 ! >>> filesink >>> location=file1.dat >>> >>> and then >>> >>> ... ! decoder ! video/x-raw-yuv- >>> strided >>> ,format >>> =(fourcc)YUY2,width=320,height=240,rowstride=700,framerate=30/1 ! >>> stridetransform ! video/x-raw- >>> yuv,format=(fourcc)YUY2,width=320,height=240,framerate=30/1 ! >>> filesink >>> location=file2.dat >>> >>> >>> But if you think this should be part of a normal pipeline, then I >>> think it would make sense to merge in with the colorspace >>> conversion, >>> so you have one memory copy instead of two. >>> >>> But do you think this is required? At least in the cases that I >>> have >>> in mind, the video sink will also support non-strided buffers, but >>> will be falling back to a less optimal mechanism. >>> >>> >>> ------ >>> >>> btw, at the risk of starting a bikeshed discussion, is 'video/x-raw- >>> yuv-full' preferred to 'video/x-raw-yuv-strided'? So far I've been >>> using the latter, but I don't mind changing the code that I've >>> written >>> so far. Should the functions added to video.c have the _full suffix >>> instead of _strided (ie. gst_video_format_parse_caps_full() instead >>> of >>> gst_video_format_parse_caps_strided())? >>> >>> If we go with -full, are there any other fields we should add to the >>> caps at the same time? Offhand, the only thing I can think of that >>> is >>> a mandatory field would be rowstride, but I haven't thought too much >>> about, for example, interlaced buffers. >>> >>> ------ >>> >>> fyi, I've begun making some changes in my private tree at >>> http://github.com/robclark/gst-plugins-base >>> ... but nothing that I've changed so far is set in stone. But we do >>> have to start adding support in the codec and other elements that >>> we'll use pretty soon, so comments and suggestions now are greatly >>> appreciated. We can stay on our own tree, or a special rowstride >>> branch, for now if integration to master is post-0.10. But it would >>> be nice to not have to *completely* re-write things later ;-) >>> >>> >>> >>> BR, >>> -R > > -------------- next part -------------- A non-text attachment was scrubbed... Name: 0001-make-gst_pad_fixate_caps-return-fixed-caps-even-if-t.patch Type: application/octet-stream Size: 1484 bytes Desc: not available URL: From francis.meyvis at gmail.com Thu Sep 10 07:46:11 2009 From: francis.meyvis at gmail.com (franchan) Date: Thu, 10 Sep 2009 07:46:11 +0200 Subject: [gst-devel] audio playback shift in time In-Reply-To: <8456544a0909040823g6e65869fx1201bab2a11421bc@mail.gmail.com> References: <8456544a0909040823g6e65869fx1201bab2a11421bc@mail.gmail.com> Message-ID: <8456544a0909092246vd56a30ax2008fd1355d79c84@mail.gmail.com> Hello gstreamer developers, Nobody knows an answer to my earlier question? Thanks, francis On Fri, Sep 4, 2009 at 5:23 PM, franchan wrote: > Hello, > > I have a custom mpegts demux. It gets data through the appsrc. > Into the appsrc I push the mpegts coming from an UDP socket. > This allows to do changing from UDP socket (other mpegts stream). > > When this happens I change the playbin2 first from playing to ready, > Then back to playing again. > > The mpegts demux, when it sees the state change, > removes its pads and waits for the new PAT/PMT information. > When the mpegts sees the new PAT PMT, it recreates the new pads. > > Before the mpegts plugin sends out the PES data, > I first send a flush-start, flush-stop and the new_segment event downstream. > with the start and position time corresponding to the first PTS > detected in the new mpegts data. > > For the first socket, when starting playback, all works fine: audio > and video are in sync. > When switching the socket the video starts playing fine after 1 to 2 seconds. > > However the audio behaves odd: its playback is delayed > by the amount of time socket 1 has been playing. > > I verified the PTS timestamps on the input (in mpegts demux) and > the output (audio/video decoders). These all seem fine; > audio/video data is alternately push forward into the sinks. > > Still the alsa sink (or something else after the audio decoder) > introduces this strange delay. > > Can somebody tell me what is going on and how to remove this delay for audio > and still having the video play back as well? > > Thanks, > francis > From francis.meyvis at gmail.com Thu Sep 10 07:47:42 2009 From: francis.meyvis at gmail.com (franchan) Date: Thu, 10 Sep 2009 07:47:42 +0200 Subject: [gst-devel] audio playback shift in time In-Reply-To: <8456544a0909040823g6e65869fx1201bab2a11421bc@mail.gmail.com> References: <8456544a0909040823g6e65869fx1201bab2a11421bc@mail.gmail.com> Message-ID: <8456544a0909092247s3791ecb2mfea6055cfdb62c47@mail.gmail.com> Hello gstreamer developers, Nobody knows an answer to my earlier question? On Fri, Sep 4, 2009 at 5:23 PM, franchan wrote: > Hello, > > I have a custom mpegts demux. It gets data through the appsrc. > Into the appsrc I push the mpegts coming from an UDP socket. > This allows to do changing from UDP socket (other mpegts stream). > > When this happens I change the playbin2 first from playing to ready, > Then back to playing again. > > The mpegts demux, when it sees the state change, > removes its pads and waits for the new PAT/PMT information. > When the mpegts sees the new PAT PMT, it recreates the new pads. > > Before the mpegts plugin sends out the PES data, > I first send a flush-start, flush-stop and the new_segment event downstream. > with the start and position time corresponding to the first PTS > detected in the new mpegts data. > > For the first socket, when starting playback, all works fine: audio > and video are in sync. > When switching the socket the video starts playing fine after 1 to 2 seconds. > > However the audio behaves odd: its playback is delayed > by the amount of time socket 1 has been playing. > > I verified the PTS timestamps on the input (in mpegts demux) and > the output (audio/video decoders). These all seem fine; > audio/video data is alternately push forward into the sinks. > > Still the alsa sink (or something else after the audio decoder) > introduces this strange delay. > > Can somebody tell me what is going on and how to remove this delay for audio > and still having the video play back as well? (sorry I forgot to include my question again ...) Thanks, francis From wim.taymans at gmail.com Thu Sep 10 08:59:56 2009 From: wim.taymans at gmail.com (Wim Taymans) Date: Thu, 10 Sep 2009 08:59:56 +0200 Subject: [gst-devel] audio playback shift in time In-Reply-To: <8456544a0909092246vd56a30ax2008fd1355d79c84@mail.gmail.com> References: <8456544a0909040823g6e65869fx1201bab2a11421bc@mail.gmail.com> <8456544a0909092246vd56a30ax2008fd1355d79c84@mail.gmail.com> Message-ID: <1252565996.11128.44.camel@metal> On Thu, 2009-09-10 at 07:46 +0200, franchan wrote: > Hello gstreamer developers, > > Nobody knows an answer to my earlier question? It could be this bug: http://bugzilla.gnome.org/show_bug.cgi?id=594136 Wim > > Thanks, > francis > > > On Fri, Sep 4, 2009 at 5:23 PM, franchan wrote: > > Hello, > > > > I have a custom mpegts demux. It gets data through the appsrc. > > Into the appsrc I push the mpegts coming from an UDP socket. > > This allows to do changing from UDP socket (other mpegts stream). > > > > When this happens I change the playbin2 first from playing to ready, > > Then back to playing again. > > > > The mpegts demux, when it sees the state change, > > removes its pads and waits for the new PAT/PMT information. > > When the mpegts sees the new PAT PMT, it recreates the new pads. > > > > Before the mpegts plugin sends out the PES data, > > I first send a flush-start, flush-stop and the new_segment event downstream. > > with the start and position time corresponding to the first PTS > > detected in the new mpegts data. > > > > For the first socket, when starting playback, all works fine: audio > > and video are in sync. > > When switching the socket the video starts playing fine after 1 to 2 seconds. > > > > However the audio behaves odd: its playback is delayed > > by the amount of time socket 1 has been playing. > > > > I verified the PTS timestamps on the input (in mpegts demux) and > > the output (audio/video decoders). These all seem fine; > > audio/video data is alternately push forward into the sinks. > > > > Still the alsa sink (or something else after the audio decoder) > > introduces this strange delay. > > > > Can somebody tell me what is going on and how to remove this delay for audio > > and still having the video play back as well? > > > > Thanks, > > francis > > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From lfarkas at lfarkas.org Thu Sep 10 10:50:05 2009 From: lfarkas at lfarkas.org (Farkas Levente) Date: Thu, 10 Sep 2009 10:50:05 +0200 Subject: [gst-devel] timestamps on a live h264 source In-Reply-To: References: Message-ID: <4AA8BDBD.9050806@lfarkas.org> hi, is there anybody how has the required knowledge about gstreamer, encoding and h264 to answer this question? since it's also big problem for us:-( On 09/08/2009 04:28 PM, PALFFY Daniel wrote: > > Hi, > > I'm developing a gstreamer source for a raw-yuv/h264 capable video grabber > card. In raw mode, the source works fine without setting > GST_BUFFER_OFFSET, GST_BUFFER_OFFSET_END, GST_BUFFER_TIMESTAMP and > GST_BUFFER_DURATION, but for h264 live play, I can't find a working > combination. > > The example pipeline looks like this: > gst-launch mysource ! "video/x-h264,framerate=25/1" ! ffdec_h264 ! xvimagesink > > The card provides each frame as a separate buffer, and (in the current > configuration) I have one SPS, one PPS, one I, and 14 P-frames in a group, > each output in a separate GstBuffer; > > When not setting anything, the pipeline takes all grabbed frames, but > displays only the first (or maybe first few). > > If I set all the values to what i believe is correct (put a serial number > incrementing from 0 in OFFSET, OFFSET+1 in OFFSET_END, a > hardware-generated timestamp in TIMESTAMP, and 0 for SPS/PPS frames > and GST_SECOND/framerate for I/P frames in DURATION), the pipeline only > takes and displays the first four frames and then stalls. > > If I count the SPS/PPS frames as normal frames, use the same duration for > them as I/P frmaes and increment the timestamp accordingly, the buffer in > my element fills slowly as the decoder takes fewer frames than produced. > > When saving the stream to a file and playing back from there, everything > works fine. > > What would be the correct values for the timestamps in this case? Or do I > have to implement a clock-capable element? -- Levente "Si vis pacem para bellum!" From wim.taymans at gmail.com Thu Sep 10 11:46:08 2009 From: wim.taymans at gmail.com (Wim Taymans) Date: Thu, 10 Sep 2009 11:46:08 +0200 Subject: [gst-devel] timestamps on a live h264 source In-Reply-To: <4AA8BDBD.9050806@lfarkas.org> References: <4AA8BDBD.9050806@lfarkas.org> Message-ID: <1252575968.4230.2.camel@metal> On Thu, 2009-09-10 at 10:50 +0200, Farkas Levente wrote: > hi, > is there anybody how has the required knowledge about gstreamer, > encoding and h264 to answer this question? since it's also big problem > for us:-( You are probably dealing with a live source and so you should use the running-time of the pipeline to timestamp outgoing buffers. Why don't we talk about it on IRC, much easier. Wim > > On 09/08/2009 04:28 PM, PALFFY Daniel wrote: > > > > Hi, > > > > I'm developing a gstreamer source for a raw-yuv/h264 capable video grabber > > card. In raw mode, the source works fine without setting > > GST_BUFFER_OFFSET, GST_BUFFER_OFFSET_END, GST_BUFFER_TIMESTAMP and > > GST_BUFFER_DURATION, but for h264 live play, I can't find a working > > combination. > > > > The example pipeline looks like this: > > gst-launch mysource ! "video/x-h264,framerate=25/1" ! ffdec_h264 ! xvimagesink > > > > The card provides each frame as a separate buffer, and (in the current > > configuration) I have one SPS, one PPS, one I, and 14 P-frames in a group, > > each output in a separate GstBuffer; > > > > When not setting anything, the pipeline takes all grabbed frames, but > > displays only the first (or maybe first few). > > > > If I set all the values to what i believe is correct (put a serial number > > incrementing from 0 in OFFSET, OFFSET+1 in OFFSET_END, a > > hardware-generated timestamp in TIMESTAMP, and 0 for SPS/PPS frames > > and GST_SECOND/framerate for I/P frames in DURATION), the pipeline only > > takes and displays the first four frames and then stalls. > > > > If I count the SPS/PPS frames as normal frames, use the same duration for > > them as I/P frmaes and increment the timestamp accordingly, the buffer in > > my element fills slowly as the decoder takes fewer frames than produced. > > > > When saving the stream to a file and playing back from there, everything > > works fine. > > > > What would be the correct values for the timestamps in this case? Or do I > > have to implement a clock-capable element? > > > From fmogollon at vicomtech.org Thu Sep 10 18:06:19 2009 From: fmogollon at vicomtech.org (Felipe =?ISO-8859-1?Q?Mogoll=F3n?=) Date: Thu, 10 Sep 2009 18:06:19 +0200 Subject: [gst-devel] Getting bin from gst-rtsp-server pipeline Message-ID: <1252598779.3340.44.camel@debian> Hi to everyone. Is there any way to get the bin from the pipeline created by gst-rtsp-server to make RTSP streaming? Thanks in advance. -- Juan Felipe Mogoll?n Rodr?guez Investigador / Researcher TV Digital y Servicios Multimedia, Digital TV and Multimedia Services, Telebista Digitala eta Multimedia zerbitzuak. Vicomtech - Visual Interaction Communication Technologies Mikeletegi Pasealekua, 57 - Parque Tecnol?gico 20009 Donostia - San Sebastin - Spain Tel: +[34] 943 30 92 30 Fax: +[34] 943 30 93 93 e-mail: fmogollon at vicomtech.org www.vicomtech.org *** member of INI-GraphicsNet **** www.inigraphics.net *** member of IK4 Research Alliance **** www.ik4.es ----------------------------------------------------- Vicomtech is an ISO 9001:2000 certified institute ----------------------------------------------------- Este mensaje se dirige exclusivamente a su destinatario. La informaci?n incluida en el presente correo es confidencial sometida a secreto profesional, especialmente en lo que respecta a los datos de car?cter personal, cuya divulgaci?n est? prohibida, en virtud de la legislaci?n vigente. Si usted no es el destinatario leg?timo y lo ha recibido por error o tiene conocimiento del mismo por cualquier motivo, le rogamos que nos lo comunique por este medio y proceda a destruirlo o borrarlo. 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From jaisena at yahoo.com Thu Sep 10 20:43:20 2009 From: jaisena at yahoo.com (jayasena s) Date: Thu, 10 Sep 2009 11:43:20 -0700 (PDT) Subject: [gst-devel] A/V sync and stutter issue In-Reply-To: <316913.58035.qm@web55304.mail.re4.yahoo.com> Message-ID: <561800.8738.qm@web55308.mail.re4.yahoo.com> ? Could someone help me in debugging and solving this issue. Thanks, Jai --- On Wed, 9/9/09, jayasena s wrote: From: jayasena s Subject: [gst-devel] A/V sync and stutter issue To: gstreamer-devel at lists.sourceforge.net Date: Wednesday, September 9, 2009, 5:43 AM Hi, ?I am having A/V Sync and stutter issues , when using totem player with gstreamer. With gst-launch, audio and video are in sync and video is smooth ( no stutter issue). ? Timestamps seem to be good, the stutter is more visible for the first 4 seconds of the playback. ? Did anyone come across this issue, Could you suggest ways to debug these issues ? Thanks, Jai -----Inline Attachment Follows----- ------------------------------------------------------------------------------ Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day trial. Simplify your report design, integration and deployment - and focus on what you do best, core application coding. Discover what's new with Crystal Reports now.? http://p.sf.net/sfu/bobj-july -----Inline Attachment Follows----- _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: From jyoti.d at allaboutif.com Fri Sep 11 11:56:36 2009 From: jyoti.d at allaboutif.com (Jyoti) Date: Fri, 11 Sep 2009 15:26:36 +0530 Subject: [gst-devel] problem disabling libxml2 in gstreamer-0.24 for TI board Message-ID: Hi all, I am compiling gstreamer-0.24 for my TI board. Though I am disabling building xml library using --disable-loadsave option I am not able to disable. The configure options I use are as below: ./configure --build=i686-linux --host=arm-linux --prefix=$PREFIX --disable-static --enable-shared --disable-libtool-lock --disable-rpath --disable-nls --disable-gst-debug --disable-loadsave --disable-trace --disable-debug --disable-profiling --disable-valgrind --disable-gcov --disable-examples --disable-tests --disable-failing-tests --disable-docbook --disable-gtk-doc --disable-alloc-trace --disable-poisoning --with-pic --without-check CFLAGS="-DG_DISABLE_ASSERT -DG_DISABLE_CHECKS -DG_DISABLE_CAST_CHECKS -I$PREFIX/include -O2" LDFLAGS=-L$PREFIX/lib and also "--enable-binary-registry" option is not seen in version gstreamer-0.10.24. Should I do any extra modifications to disable building xml. On running configure the below are messages for XML ************************************************************************************************** checking for XML... yes configure: Test xml2 program linked configure: Using /opt/gstreamer/lib/gstreamer-0.10 as the plugin install location *************************************************************************************************** Jyoti -------------- next part -------------- An HTML attachment was scrubbed... URL: From t.i.m at zen.co.uk Fri Sep 11 12:24:26 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Fri, 11 Sep 2009 11:24:26 +0100 Subject: [gst-devel] problem disabling libxml2 in gstreamer-0.24 for TI board In-Reply-To: References: Message-ID: <1252664666.4870.3.camel@zingle> On Fri, 2009-09-11 at 15:26 +0530, Jyoti wrote: > I am compiling gstreamer-0.24 for my TI board. > Though I am disabling building xml library using --disable-loadsave > option I am not able to disable. > > The configure options I use are as below: (snip) > Should I do any extra modifications to disable building xml. > > On running configure the below are messages for XML > checking for XML... yes > configure: Test xml2 program linked Sounds like you're running into https://bugzilla.gnome.org/show_bug.cgi?id=590841 The fix for this can be found here: http://cgit.freedesktop.org/gstreamer/gstreamer/commit/?id=cc57c404fd722cfe47f337b645e26197baab4c97 > and also "--enable-binary-registry" option is not seen in version > gstreamer-0.10.24. That configure option has been removed. It is not needed any longer since the binary registry is now the default (and the only option). Cheers -Tim From thaytan at noraisin.net Fri Sep 11 12:21:50 2009 From: thaytan at noraisin.net (Jan Schmidt) Date: Fri, 11 Sep 2009 11:21:50 +0100 Subject: [gst-devel] Release plan for this month: Core/Base/Python/FFmpeg Message-ID: <1252664510.5346.8.camel@fancy> Hi all, The release schedule currently says that gst-plugins-ugly and gst-ffmpeg are due for release in this month's cycle. It was pointed out though, that the GNOME 2.28 tarballs are due this month, and distros will be rolling things up shortly afterward. Accordingly, if we pull the Core/Base and Python releases forward from next month, there's a good chance that distros will be able to include them in their next release. There's a few issues in the 0.10.24 core/base that make this an attractive prospect. However, gst-ffmpeg is also in need of a release to fix a nasty bug in the last tarballs, so the plan is that Core/Base/Python *and* gst-ffmpeg will be released this month. I'll freeze all 4 modules tonight, and make pre-releases. Following a normal 2 week release cycle for the modules, that means the tarballs should be out Sept 25th. Cheers, Jan. -- Jan Schmidt From thaytan at noraisin.net Fri Sep 11 12:22:51 2009 From: thaytan at noraisin.net (Jan Schmidt) Date: Fri, 11 Sep 2009 11:22:51 +0100 Subject: [gst-devel] A/V sync and stutter issue In-Reply-To: <561800.8738.qm@web55308.mail.re4.yahoo.com> References: <561800.8738.qm@web55308.mail.re4.yahoo.com> Message-ID: <1252664571.5346.9.camel@fancy> On Thu, 2009-09-10 at 11:43 -0700, jayasena s wrote: > > Could someone help me in debugging and solving this issue. > > Thanks, > Jai > > --- On Wed, 9/9/09, jayasena s wrote: > > > From: jayasena s > Subject: [gst-devel] A/V sync and stutter issue > To: gstreamer-devel at lists.sourceforge.net > Date: Wednesday, September 9, 2009, 5:43 AM > > Hi, > I am having A/V Sync and stutter issues , when using totem > player with gstreamer. > With gst-launch, audio and video are in sync and video is > smooth ( no stutter issue). > > Timestamps seem to be good, the stutter is more visible for > the first 4 seconds of the playback. > > Did anyone come across this issue, Could you suggest ways to > debug these issues > > Thanks, > Jai > > > > -----Inline Attachment Follows----- > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal > Reports 2008 30-Day > trial. Simplify your report design, integration and deployment > - and focus on > what you do best, core application coding. Discover what's new > with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > > -----Inline Attachment Follows----- > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -- Jan Schmidt From jyoti.d at allaboutif.com Fri Sep 11 13:00:39 2009 From: jyoti.d at allaboutif.com (Jyoti) Date: Fri, 11 Sep 2009 16:30:39 +0530 Subject: [gst-devel] problem disabling libxml2 in gstreamer-0.24 for TI board In-Reply-To: <1252664666.4870.3.camel@zingle> References: <1252664666.4870.3.camel@zingle> Message-ID: Thanks a lot Tim. On Fri, Sep 11, 2009 at 3:54 PM, Tim-Philipp M?ller wrote: > On Fri, 2009-09-11 at 15:26 +0530, Jyoti wrote: > > > I am compiling gstreamer-0.24 for my TI board. > > Though I am disabling building xml library using --disable-loadsave > > option I am not able to disable. > > > > The configure options I use are as below: (snip) > > Should I do any extra modifications to disable building xml. > > > > On running configure the below are messages for XML > > checking for XML... yes > > configure: Test xml2 program linked > > Sounds like you're running into > https://bugzilla.gnome.org/show_bug.cgi?id=590841 > > The fix for this can be found here: > > http://cgit.freedesktop.org/gstreamer/gstreamer/commit/?id=cc57c404fd722cfe47f337b645e26197baab4c97 > > > > > and also "--enable-binary-registry" option is not seen in version > > gstreamer-0.10.24. > > That configure option has been removed. It is not needed any longer > since the binary registry is now the default (and the only option). > > Cheers > -Tim > > > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus > on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From otte at gnome.org Fri Sep 11 19:02:48 2009 From: otte at gnome.org (Benjamin Otte) Date: Fri, 11 Sep 2009 19:02:48 +0200 Subject: [gst-devel] Using cairo/pixman for raw video in GStreamer Message-ID: Hi, This is an idea that's been brewing in my head for a bit. After thinking about it for a while and poking some people on IRC, I'm pretty convinced it's the best way forward. Here's a list of problems I'd like to see solved: 1) Correctly identify video in the GStreamer elements (stride, width, height, size of image and components) In the short while I recently hacked on plugins, I found bugs in lots of places, from common to obscure formats. And those were in pretty common elements (theoraenc/dec, videotestsrc). Using the APIs in gstvideo pretty much solves this problem for the current set of plugins. 2) Allow drawing onto different video formats There is actually multiple issues here: For a start, elements that draw to various YUV formats often get it wrong - mostly in corner cases. Others take shortcuts that degrade the quality of the video (like videotestsrc not computing the average for U and V pixels for subsampled planes). Examples of elements doing drawing operations start with elements like videocrop, videobox that resize the input or videotestsrc that draws rectangles. Next step there's videomixer and textoverlay that compose various input streams. And then there's various effect elements like smpte or effectv or even videoscale at the top end. And almost all of these elements support only a very limited set of colorspaces - I420 and AYUV mostly. (Also, I always dreamed of doing an mplayer gstreamer filter that responds to keypresses and displays the volume/brightness etc UI on top of the video. That's really hard to do currently.) 3) Allow better interaction between applications consuming video and GStreamer This is mostly related to web browsers, but applies to Flash, Clutter, games and probably lots of other things, too: They all want to get access to the video data and do stuff with it. Currently this often involves a colorspace conversion to RGB and then stuffing that into a Cairo surface. It would be much nicer if Cairo and pixman supported YUV so the colorpsace conversion could be omitted when the hardware accepts it. The same goes in the other direction: I'd like to capture the screen as YUV, not as RGB, if I record it to Theora video. 4) Allow hw-acceleration in the video pipeline Decoding a H264 stream in hardware, rendering sutitles on top of it, scaling it to fit and displaying as fullscreen video on my computer can in theory all be done in hardware. Unfortunately, GStreamer currently lacks infrastructure for this, so all this stuff ends up being done in software. 5) Figuring out the porper format to use is an art So where do you put the conversion element? Do you even have to put one? Newcomers trip over these problems a lot and I still hate having to edit gst-launch lines because I forgot some converter element somewhere and now negotiation fails. I'd like this to happen automatically. Of course, it doesn't mean unnecessary colorspace conversions should happen, and I also should be able to force a certain format if I want to (important for testing). These are the steps I'd like to propose as a solution: 1) Add extensive YUV support to pixman The goal is to add an infrastructure so one can support at least the formats supported by ffmpegcolorspace today. In fact the ffmpeg infrastructure fits pretty well to pixman, but I'm not sure if a straight port is acceptable license-wise. 2) Add support to Cairo to create surfaces from any pixman image I'm not sure how hard this would be, as it basically circumvents cairo_format_t - might be possible to hook it into image surfaces or might be better to use a different surface backend. But it'd just add a single function like cairo_pixman_surface_create (pixman_image_t *image); 3) Add a new caps to gstreamer: video/cairo I'm not sure yet about the specific properties required, but certainly framerate, width and height are required. Probably an optional pixman-format is required, too. Buffers passed in this format only contain a reference to a cairo surface in the buffer's data. 4) Port elements to use this cairo API Either add new elements (cairovideotestsrc, cairocolorspace) or add support for the old ones. While doing this, refine and improve cairo or pixman, so the elements can be implemented as nicely as possible. A lot of code inside GStreamer should go away 5) Finalize APIs for pixman, cairo and GStreamer in unison After enough code got ported (not sure if those should be separate branches or if it should be part of experimental releases), we sit together and finalize the API. At this point GStremaer elements switch to using video/cairo as the default data passing format. 6) For next major GStreamer release, remove video/x-raw-* The old formats are not needed anymore, they can be removed. All elements are ported to the new API. I think these steps would solve most of the problems I outlined above. Of course some questions have come up about this that Id like to answer before somebody has to ask this question in here: 1) "This is never gonna be fast enough" I don't see why. Most of the operations people care about are just memcpys and pixman is very good at detecting them and making them fast. In fact, pixman has a huge infrastructure dedicated to speeding up things that GStreamer cannot match. And no, the current scarce usage of liboil doesn't count. Currently in a lot of cases unnecessary colorspace conversions cost a lot of performance and these will go away if every element supports every format. In short: I wouldn't have proposed this if I'd think it'd make stuff slower. 2) "I will have less control over what happens" No you won't. You'll be able to use the same formats as today and access their data just like today. You just use pixman functions instead of gst_video_* functions. I don't intend to move control away from developers. The goal is to make life simpler for developers, not harder. 3) "Adding new features to GStreamer will be a lot harder" This is only halfway true. You will still be able to write elements like you do today by accessing the raw data of the surface. Of course, if you want to add a new YUV format, it will require support in pixman, and this requires more work (or even depending on unstable versions of pixman). On the other hand, once pixman supports that element, all other GStreamer elements will support it automatically and you can start rendering subtitles onto it. I also do not believe that adding more formats is somehow a common thing that happens very often, so it can easily wait until the next pixman or cairo release. But yes, depending on other libraries reduces your options. 4) "Cairo/GStreamer developers will not like that" In fact, I talked to both of the maintainers and the response in both cases was pretty positive, but skeptical about the feasability of such a project, mostly fueled by preconceptions about what Cairo or GStreamer is and how it works. I consider myself part of both the Cairo and GStreamer comunities and know the code in quite some detail and I do think it's a very good fit. So, opinions, questions, encouragement or anything else? Benjamin From thaytan at noraisin.net Sat Sep 12 00:02:00 2009 From: thaytan at noraisin.net (Jan Schmidt) Date: Fri, 11 Sep 2009 23:02:00 +0100 Subject: [gst-devel] A/V sync and stutter issue In-Reply-To: <1252664571.5346.9.camel@fancy> References: <561800.8738.qm@web55308.mail.re4.yahoo.com> <1252664571.5346.9.camel@fancy> Message-ID: <1252706520.13334.1.camel@fancy> Hi, On Fri, 2009-09-11 at 11:22 +0100, Jan Schmidt wrote: Hrmn, my initial reply was lacking somewhat in detail ;) > On Thu, 2009-09-10 at 11:43 -0700, jayasena s wrote: > > *snip* > > Hi, > > I am having A/V Sync and stutter issues , when using totem > > player with gstreamer. > > With gst-launch, audio and video are in sync and video is > > smooth ( no stutter issue). > > > > Timestamps seem to be good, the stutter is more visible for > > the first 4 seconds of the playback. > > > > Did anyone come across this issue, Could you suggest ways to > > debug these issues I'd start by identifying the difference between the pipeline you're giving gst-launch and what totem is using. Totem (depending on the version) is using either playbin or playbin2, with the gconf audio and video sinks. Cheers, Jan. > > > > Thanks, > > Jai > > > > > > > > -----Inline Attachment Follows----- > > > > ------------------------------------------------------------------------------ > > Let Crystal Reports handle the reporting - Free Crystal > > Reports 2008 30-Day > > trial. Simplify your report design, integration and deployment > > - and focus on > > what you do best, core application coding. Discover what's new > > with > > Crystal Reports now. http://p.sf.net/sfu/bobj-july > > > > -----Inline Attachment Follows----- > > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > ------------------------------------------------------------------------------ > > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > > trial. Simplify your report design, integration and deployment - and focus on > > what you do best, core application coding. Discover what's new with > > Crystal Reports now. http://p.sf.net/sfu/bobj-july > > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -- Jan Schmidt From ds at entropywave.com Sat Sep 12 00:11:26 2009 From: ds at entropywave.com (David Schleef) Date: Fri, 11 Sep 2009 15:11:26 -0700 Subject: [gst-devel] Using cairo/pixman for raw video in GStreamer In-Reply-To: References: Message-ID: <20090911221126.GA27449@entropywave.com> On Fri, Sep 11, 2009 at 07:02:48PM +0200, Benjamin Otte wrote: > Hi, > > This is an idea that's been brewing in my head for a bit. After > thinking about it for a while and poking some people on IRC, I'm > pretty convinced it's the best way forward. > > Here's a list of problems I'd like to see solved: > > 1) Correctly identify video in the GStreamer elements (stride, width, > height, size of image and components) > In the short while I recently hacked on plugins, I found bugs in lots > of places, from common to obscure formats. And those were in pretty > common elements (theoraenc/dec, videotestsrc). Using the APIs in > gstvideo pretty much solves this problem for the current set of > plugins. Just because an element is common doesn't mean it's been reviewed in several years. videotestsrc predates gstvideo, and hasn't been updated to use gstvideo. It should. But I've been waiting until I've moved some additional video frame stuff from Cog/Schroedinger into GStreamer. The cog/schro is well-tested and calculation of frame component sizes is more obvious than in videotestsrc. Also, it supports more formats. > 2) Allow drawing onto different video formats > There is actually multiple issues here: For a start, elements that > draw to various YUV formats often get it wrong - mostly in corner > cases. Others take shortcuts that degrade the quality of the video > (like videotestsrc not computing the average for U and V pixels for > subsampled planes). See cog. > 4) Allow hw-acceleration in the video pipeline > Decoding a H264 stream in hardware, rendering sutitles on top of it, > scaling it to fit and displaying as fullscreen video on my computer > can in theory all be done in hardware. Unfortunately, GStreamer > currently lacks infrastructure for this, so all this stuff ends up > being done in software. What can be implemented has been implemented in gst-plugins-gl, and works relatively well. It needs to be connected with VDPAU/VAAPI, but that's something that still requires work at a lower level. dave... From thaytan at noraisin.net Sat Sep 12 02:17:00 2009 From: thaytan at noraisin.net (Jan Schmidt) Date: Sat, 12 Sep 2009 01:17:00 +0100 Subject: [gst-devel] Core/Base/Python/FFmpeg - 1st pre-releases Message-ID: <1252714620.13334.38.camel@fancy> Hi all, All four of the modules in this release cycle are now frozen for commits, and the 1st pre-releases are available. The tarballs are available at: http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.24.2.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.24.2.tar.bz2 http://gstreamer.freedesktop.org/src/gst-python/pre/gst-python-0.10.16.2.tar.bz2 http://gstreamer.freedesktop.org/src/gst-ffmpeg/pre/gst-ffmpeg-0.10.8.2.tar.bz2 md5sums: 851fe968803b8a49fdac513435142f0b gstreamer-0.10.24.2.tar.bz2 05cd5878ad4182c803c821cd053fdcce gst-plugins-base-0.10.24.2.tar.bz2 e513c18c22665146eca98a2dd390b42a gst-python-0.10.16.2.tar.bz2 e6057902f39301eb2d08d48b34a91dd6 gst-ffmpeg-0.10.8.2.tar.bz2 Please give them a good test and report any problems, and especially regressions, in bugzilla - http://bugzilla.gnome.org/ See http://gstreamer.freedesktop.org/wiki/ReleasePlanning2009-2 for more details about the release schedule. Cheers, Jan. -- Jan Schmidt From ping.gao at harman.com Sat Sep 12 04:19:48 2009 From: ping.gao at harman.com (Gao, Ping) Date: Fri, 11 Sep 2009 21:19:48 -0500 Subject: [gst-devel] youtube playback applicaton using gstreamer elements Message-ID: <6DF262F0BAB5944AB03D1FE91354A1EF05FFD8FDB6@HICGWSEX01.ad.harman.com> Hi there: I am trying to enable youtube playback on my target device. I have gstreamer-0.10 compiled and audio and video file playback working. How to make Youtube playback work? What elements are required? How does the pipeline look like? Thanks a lot. Ping -------------- next part -------------- An HTML attachment was scrubbed... URL: From sebastian.droege at collabora.co.uk Sat Sep 12 11:11:38 2009 From: sebastian.droege at collabora.co.uk (Sebastian =?ISO-8859-1?Q?Dr=F6ge?=) Date: Sat, 12 Sep 2009 11:11:38 +0200 Subject: [gst-devel] Using cairo/pixman for raw video in GStreamer In-Reply-To: References: Message-ID: <1252746698.6344.599.camel@odin.lan> Am Freitag, den 11.09.2009, 19:02 +0200 schrieb Benjamin Otte: > These are the steps I'd like to propose as a solution: > > 1) Add extensive YUV support to pixman > The goal is to add an infrastructure so one can support at least the > formats supported by ffmpegcolorspace today. In fact the ffmpeg > infrastructure fits pretty well to pixman, but I'm not sure if a > straight port is acceptable license-wise. > > 2) Add support to Cairo to create surfaces from any pixman image > I'm not sure how hard this would be, as it basically circumvents > cairo_format_t - might be possible to hook it into image surfaces or > might be better to use a different surface backend. But it'd just add > a single function like cairo_pixman_surface_create (pixman_image_t > *image); As you can get a cairo_t from every surface, would this mean that all cairo stuff has to support all the pixman surface formats? I mean, would it be possible later to set YUV colors instead of RGB colors in cairo, would it be possible to draw lines or whatever in cairo on some YUV surface, etc? In that case you probably have a lot of work to do ;) > 3) Add a new caps to gstreamer: video/cairo > I'm not sure yet about the specific properties required, but certainly > framerate, width and height are required. Probably an optional > pixman-format is required, too. Buffers passed in this format only > contain a reference to a cairo surface in the buffer's data. (Should really be video/x-cairo). The properties should probably include the pixman-format too because this way elements can still give preferences (if they work really better on some format than on another) or if elements can still only work on a single format (because they need to fiddle with the bits themself instead of using cairo). > 4) Port elements to use this cairo API > Either add new elements (cairovideotestsrc, cairocolorspace) or add > support for the old ones. While doing this, refine and improve cairo > or pixman, so the elements can be implemented as nicely as possible. A > lot of code inside GStreamer should go away That would be the same as gst-plugins-gl works nowadays. I'm all for it, that's definitely a good idea. Not sure if we want a cairo dependency on every video element now already though... > 6) For next major GStreamer release, remove video/x-raw-* > The old formats are not needed anymore, they can be removed. All > elements are ported to the new API. Which means that cairo/pixman must have a good framework in place to also add new formats easily. If that's given it might make sense, yes. cairo/pixman should also support 8 bit and 16 bit grayscale and the different Bayer formats too then btw. Also it would mean, that if you have some codec that decodes into some weird colorformat that is not supported by pixman/cairo yet, that you need to wait for pixman/cairo to support it and gst-plugins-foo to depend on that version or that you have to do conversions internally. > 4) "Cairo/GStreamer developers will not like that" > In fact, I talked to both of the maintainers and the response in both > cases was pretty positive, but skeptical about the feasability of such > a project, mostly fueled by preconceptions about what Cairo or > GStreamer is and how it works. I consider myself part of both the > Cairo and GStreamer comunities and know the code in quite some detail > and I do think it's a very good fit. Until step 4 of your plan it definitely makes sense and should be done. Start a GIT repository for gst-plugins-cairo and I'd help you to get the GStreamer part of things done ;) This could also be done today with just supporting RGB/RGBA. All other steps might not make sense not sure what the cairo/pixman people think about supporting random color formats. Also it would mean that GStreamer depends on cairo as a required dependency. But I guess cairo/pixman are at least portable enough to work everywhere. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Dies ist ein digital signierter Nachrichtenteil URL: From sebastian.droege at collabora.co.uk Sat Sep 12 11:13:32 2009 From: sebastian.droege at collabora.co.uk (Sebastian =?ISO-8859-1?Q?Dr=F6ge?=) Date: Sat, 12 Sep 2009 11:13:32 +0200 Subject: [gst-devel] Using cairo/pixman for raw video in GStreamer In-Reply-To: References: Message-ID: <1252746812.6344.601.camel@odin.lan> Am Freitag, den 11.09.2009, 19:02 +0200 schrieb Benjamin Otte: > [...] > 6) For next major GStreamer release, remove video/x-raw-* > The old formats are not needed anymore, they can be removed. All > elements are ported to the new API. > [...] Oh another thing. Maybe only make video/x-cairo the prefered option but keep video/x-raw-* as an option. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Dies ist ein digital signierter Nachrichtenteil URL: From miquel.farre at gmail.com Sat Sep 12 11:15:18 2009 From: miquel.farre at gmail.com (=?ISO-8859-1?Q?Miquel_=C0ngel_Farr=E9?=) Date: Sat, 12 Sep 2009 11:15:18 +0200 Subject: [gst-devel] libtool, automake and C++ library Message-ID: <49c4aa740909120215oc6ff5e9k3e3a061ad9b2c86e@mail.gmail.com> Good Morning, Just asked on the irc channel and MikeS-tp tried to help (thanks!) .. this is the problem: I am trying to build a C++ plugin, flags and libs are given correctly nevertheless It doesen't build because it is C++ code, I have arrived to this conclusion making a little test: - Try to build gst-template/gst-plugin: ./autogen.sh && make : it works - Change gstplugin.c to gstplugin.cpp edit Makefile.am with the new file extension - autogen.sh && make result: /bin/bash ../libtool --tag=CXX --tag=disable-static --mode=compile g++ -DHAVE_CONFIG_H -I. -I.. -g -O2 -MT libgstplugin_la-gstplugin.lo -MD -MP -MF .deps/libgstplugin_la-gstplugin.Tpo -c -o libgstplugin_la-gstplugin.lo `test -f 'gstplugin.cpp' || echo './'`gstplugin.cpp libtool: compile: g++ -DHAVE_CONFIG_H -I. -I.. -g -O2 -MT libgstplugin_la-gstplugin.lo -MD -MP -MF .deps/libgstplugin_la-gstplugin.Tpo -c gstplugin.cpp -fPIC -DPIC -o .libs/libgstplugin_la-gstplugin.o gstplugin.cpp:63:21: error: gst/gst.h: No such file or directory In file included from gstplugin.cpp:65: gstplugin.h:65: error: expected constructor, destructor, or type conversion before 'typedef' gstplugin.h:70: error: 'GstElement' does not name a type gstplugin.h:72: error: ISO C++ forbids declaration of 'GstPad' with no type gstplugin.h:72: error: expected ';' before '*' token gstplugin.h:74: error: 'gboolean' does not name a type . . . I have tryed to add params to libtool like --tag=CXX.. Anybody knows what and where params should I add? Thanks a lot -------------- next part -------------- An HTML attachment was scrubbed... URL: From sebastian.droege at collabora.co.uk Sat Sep 12 11:18:01 2009 From: sebastian.droege at collabora.co.uk (Sebastian =?ISO-8859-1?Q?Dr=F6ge?=) Date: Sat, 12 Sep 2009 11:18:01 +0200 Subject: [gst-devel] libtool, automake and C++ library In-Reply-To: <49c4aa740909120215oc6ff5e9k3e3a061ad9b2c86e@mail.gmail.com> References: <49c4aa740909120215oc6ff5e9k3e3a061ad9b2c86e@mail.gmail.com> Message-ID: <1252747081.6344.602.camel@odin.lan> Am Samstag, den 12.09.2009, 11:15 +0200 schrieb Miquel ?ngel Farr?: > [...] > I have tryed to add params to libtool like --tag=CXX.. > > Anybody knows what and where params should I add? I don't know right now but you might want to take a look at the soundtouch or modplug plugins in gst-plugins-bad. They're written in C++ too. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Dies ist ein digital signierter Nachrichtenteil URL: From msclrhd at googlemail.com Sat Sep 12 11:21:58 2009 From: msclrhd at googlemail.com (Reece Dunn) Date: Sat, 12 Sep 2009 10:21:58 +0100 Subject: [gst-devel] libtool, automake and C++ library In-Reply-To: <49c4aa740909120215oc6ff5e9k3e3a061ad9b2c86e@mail.gmail.com> References: <49c4aa740909120215oc6ff5e9k3e3a061ad9b2c86e@mail.gmail.com> Message-ID: <3f4fd2640909120221k399a9967uc3ab11c8627d68c3@mail.gmail.com> 2009/9/12 Miquel ?ngel Farr? : > Good Morning, > > Just asked on the irc channel and MikeS-tp tried to help (thanks!) .. this > is the problem: > > I am trying to build a C++ plugin, flags and libs are given correctly > nevertheless It doesen't build because it is C++ code, I have arrived to > this conclusion making a little test: > > - Try to build gst-template/gst-plugin: ./autogen.sh && make : it works > - Change gstplugin.c to gstplugin.cpp edit Makefile.am with the new file > extension > - autogen.sh && make result: > > gstplugin.cpp:63:21: error: gst/gst.h: No such file or directory Have you set CXXFLAGS to contain the include path instead of CFLAGS? - Reece From miquel.farre at gmail.com Sat Sep 12 11:30:03 2009 From: miquel.farre at gmail.com (=?ISO-8859-1?Q?Miquel_=C0ngel_Farr=E9?=) Date: Sat, 12 Sep 2009 11:30:03 +0200 Subject: [gst-devel] libtool, automake and C++ library In-Reply-To: <3f4fd2640909120221k399a9967uc3ab11c8627d68c3@mail.gmail.com> References: <49c4aa740909120215oc6ff5e9k3e3a061ad9b2c86e@mail.gmail.com> <3f4fd2640909120221k399a9967uc3ab11c8627d68c3@mail.gmail.com> Message-ID: <49c4aa740909120230t1c6c6920ve22bb5924a8808e@mail.gmail.com> Reece, Yes I did it, with the same problem, these are my flags: libgstplugin_la_CFLAGS = $(GST_CXXFLAGS) libgstplugin_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS) $(GSTCTRL_LIBS) libgstplugin_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) libgstplugin_la_LIBTOOLFLAGS = --tag=disable-static Nevertheless looking at the generated makefile there is nothing for GST_CXXFLAGS, should I add something to configure.ac? Thanks 2009/9/12 Reece Dunn > 2009/9/12 Miquel ?ngel Farr? : > > Good Morning, > > > > Just asked on the irc channel and MikeS-tp tried to help (thanks!) .. > this > > is the problem: > > > > I am trying to build a C++ plugin, flags and libs are given correctly > > nevertheless It doesen't build because it is C++ code, I have arrived to > > this conclusion making a little test: > > > > - Try to build gst-template/gst-plugin: ./autogen.sh && make : it works > > - Change gstplugin.c to gstplugin.cpp edit Makefile.am with the new file > > extension > > - autogen.sh && make result: > > > > gstplugin.cpp:63:21: error: gst/gst.h: No such file or directory > > Have you set CXXFLAGS to contain the include path instead of CFLAGS? > > - Reece > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus > on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From sebastian.droege at collabora.co.uk Sat Sep 12 11:41:00 2009 From: sebastian.droege at collabora.co.uk (Sebastian =?ISO-8859-1?Q?Dr=F6ge?=) Date: Sat, 12 Sep 2009 11:41:00 +0200 Subject: [gst-devel] libtool, automake and C++ library In-Reply-To: <49c4aa740909120230t1c6c6920ve22bb5924a8808e@mail.gmail.com> References: <49c4aa740909120215oc6ff5e9k3e3a061ad9b2c86e@mail.gmail.com> <3f4fd2640909120221k399a9967uc3ab11c8627d68c3@mail.gmail.com> <49c4aa740909120230t1c6c6920ve22bb5924a8808e@mail.gmail.com> Message-ID: <1252748460.6344.604.camel@odin.lan> Am Samstag, den 12.09.2009, 11:30 +0200 schrieb Miquel ?ngel Farr?: > Reece, > > Yes I did it, with the same problem, these are my flags: > > libgstplugin_la_CFLAGS = $(GST_CXXFLAGS) > libgstplugin_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS) $(GSTCTRL_LIBS) > libgstplugin_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) > libgstplugin_la_LIBTOOLFLAGS = --tag=disable-static > > Nevertheless looking at the generated makefile there is nothing for > GST_CXXFLAGS, should I add something to configure.ac? It should be the other way around: libgstplugin_la_CXXFLAGS = $(GST_CFLAGS) -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Dies ist ein digital signierter Nachrichtenteil URL: From miquel.farre at gmail.com Sat Sep 12 11:46:24 2009 From: miquel.farre at gmail.com (=?ISO-8859-1?Q?Miquel_=C0ngel_Farr=E9?=) Date: Sat, 12 Sep 2009 11:46:24 +0200 Subject: [gst-devel] libtool, automake and C++ library In-Reply-To: <1252748460.6344.604.camel@odin.lan> References: <49c4aa740909120215oc6ff5e9k3e3a061ad9b2c86e@mail.gmail.com> <3f4fd2640909120221k399a9967uc3ab11c8627d68c3@mail.gmail.com> <49c4aa740909120230t1c6c6920ve22bb5924a8808e@mail.gmail.com> <1252748460.6344.604.camel@odin.lan> Message-ID: <49c4aa740909120246x7509f102s1316616e6bb37c81@mail.gmail.com> solved, thanks guys!! 2009/9/12 Sebastian Dr?ge > Am Samstag, den 12.09.2009, 11:30 +0200 schrieb Miquel ?ngel Farr?: > > Reece, > > > > Yes I did it, with the same problem, these are my flags: > > > > libgstplugin_la_CFLAGS = $(GST_CXXFLAGS) > > libgstplugin_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS) $(GSTCTRL_LIBS) > > libgstplugin_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) > > libgstplugin_la_LIBTOOLFLAGS = --tag=disable-static > > > > Nevertheless looking at the generated makefile there is nothing for > > GST_CXXFLAGS, should I add something to configure.ac? > > It should be the other way around: > > libgstplugin_la_CXXFLAGS = $(GST_CFLAGS) > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus > on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From msclrhd at googlemail.com Sat Sep 12 11:49:41 2009 From: msclrhd at googlemail.com (Reece Dunn) Date: Sat, 12 Sep 2009 10:49:41 +0100 Subject: [gst-devel] libtool, automake and C++ library In-Reply-To: <49c4aa740909120230t1c6c6920ve22bb5924a8808e@mail.gmail.com> References: <49c4aa740909120215oc6ff5e9k3e3a061ad9b2c86e@mail.gmail.com> <3f4fd2640909120221k399a9967uc3ab11c8627d68c3@mail.gmail.com> <49c4aa740909120230t1c6c6920ve22bb5924a8808e@mail.gmail.com> Message-ID: <3f4fd2640909120249m5afaee65va1d4fecf8ca988b1@mail.gmail.com> 2009/9/12 Miquel ?ngel Farr? : > Reece, > > ?Yes I did it, with the same problem, these are my flags: > > libgstplugin_la_CFLAGS = $(GST_CXXFLAGS) Try: libgstplugin_la_CXXFLAGS = $(GST_CFLAGS) > libgstplugin_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS) $(GSTCTRL_LIBS) > libgstplugin_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) > libgstplugin_la_LIBTOOLFLAGS = --tag=disable-static > > Nevertheless looking at the generated makefile there is nothing for > GST_CXXFLAGS, should I add something to configure.ac? The CFLAGS variable is used for C sources, and CXXFLAGS for C++ sources. The libgstplugin_la_* variables apply to the libgstplugin.la target. You already have: PKG_CHECK_MODULES([GST], [...]) AC_SUBST(GST_CFLAGS) AC_SUBST(GST_LIBS) (as it works with the C sources) so you don't need to change anything there. NOTE: Since the GStreamer library is a C library, it generates a CFLAGS variable. The PKG_CHECK_MODULES prepends the first argument to it with an _ at the end (GST in the above example). The AC_SUBST(variable) searches for all occurrances of variable in the source files for AC_CONFIG_FILES and replaces it with the content if the variable. - Reece From slomo at circular-chaos.org Sat Sep 12 11:40:41 2009 From: slomo at circular-chaos.org (Sebastian =?ISO-8859-1?Q?Dr=F6ge?=) Date: Sat, 12 Sep 2009 11:40:41 +0200 Subject: [gst-devel] libtool, automake and C++ library In-Reply-To: <49c4aa740909120230t1c6c6920ve22bb5924a8808e@mail.gmail.com> References: <49c4aa740909120215oc6ff5e9k3e3a061ad9b2c86e@mail.gmail.com> <3f4fd2640909120221k399a9967uc3ab11c8627d68c3@mail.gmail.com> <49c4aa740909120230t1c6c6920ve22bb5924a8808e@mail.gmail.com> Message-ID: <1252748441.6344.603.camel@odin.lan> Am Samstag, den 12.09.2009, 11:30 +0200 schrieb Miquel ?ngel Farr?: > Reece, > > Yes I did it, with the same problem, these are my flags: > > libgstplugin_la_CFLAGS = $(GST_CXXFLAGS) > libgstplugin_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS) $(GSTCTRL_LIBS) > libgstplugin_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) > libgstplugin_la_LIBTOOLFLAGS = --tag=disable-static > > Nevertheless looking at the generated makefile there is nothing for > GST_CXXFLAGS, should I add something to configure.ac? It should be the other way around: libgstplugin_la_CXXFLAGS = $(GST_CFLAGS) -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Dies ist ein digital signierter Nachrichtenteil URL: From ensonic at hora-obscura.de Sat Sep 12 12:32:37 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Sat, 12 Sep 2009 13:32:37 +0300 Subject: [gst-devel] Naming threads ? Other performance debugging helpers ? In-Reply-To: <1efe3a6e0909082318i1581b7dt3e9afce6ec6620ce@mail.gmail.com> References: <1efe3a6e0909082318i1581b7dt3e9afce6ec6620ce@mail.gmail.com> Message-ID: <4AAB78C5.4080304@hora-obscura.de> Hi Florent, Florent schrieb: > Hi > > I was wondering if it would be feasible that elements that launch > threads (e.g. queues) could name threads (using prctl(PR_SET_NAME? ), > with the name being constructed after their name, or the neighbour > elements' name. After a quick search [1], i'm not sure the Linux > kernel offers this feature https://bugzilla.gnome.org/show_bug.cgi?id=580505 I'd like to have this too, but I don't think it can be done in a crossplatform, non hackish way. > > This would help a lot performance debugging : so far if you want to > know what thread is consuming the most power, you make assumptions on > the nature of processing, and study partial pipelines to estimate the > proportion taken by what, but this approach is very limiting for > detecting/understanding side effects. What you could do is to add some thread-id name association to gstreamer. The you could write gdb extension to query it? Or the logging could print the name as well. Where would you need the name? > > Also, any other idea about how to "map" the CPU cycles consumption of > a pipeline or other performance indicators would be welcome. I am > considering developing a tool which would display states/hints of a > pipeline's behaviour in a graphical fashion. So far i experimented > with polling queue filling states, but it does not always prove > helpful. I also don't know how i can access QoS data from python. As my time permits I hack on gst-tracelib [1]. One plan is to intergrate the graphing (which is done by gnuplot) into gst-debug-viewer [2]. The later is written in python. WOuld be cool if you would consider helping on that one. Stefan [1] http://cgit.freedesktop.org/~ensonic/gst-tracelib/ [2] http://cgit.freedesktop.org/~cymacs/gst-debug-viewer/ > > Regards, > > FLorent > > [1] http://www.gossamer-threads.com/lists/engine?list=linux&do=search_results&search_forum=forum_1&search_string=thread+name&search_type=AND > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From bilboed at gmail.com Sat Sep 12 13:19:43 2009 From: bilboed at gmail.com (Edward Hervey) Date: Sat, 12 Sep 2009 13:19:43 +0200 Subject: [gst-devel] [ANNOUNCE] PiTiVi 0.13.3 "... we shall never (sur)render" Message-ID: <1252754383.2317.19.camel@putamadre> This mail announces the release of PiTiVi video editor 0.13.3 "... we shall never (sur)render". PiTiVi is an open source video editor, written in Python and based on GStreamer and GTK+. More information in the attached release notes, as well as on http://www.pitivi.org/ To file bugs, please go to http://bugzilla.gnome.org/enter_bug.cgi?product=pitivi -------------- next part -------------- 0.13.3 Release "... we shall never (sur)render" ----------------------------------------------- The PiTiVi team is proud to announce the third release in the unstable 0.13 PiTiVi series. Due to its dependency on GStreamer, The PiTiVi team strongly recommends users have all official latest gstreamer libraries and plugins installed for the best user experience. Title is from a quote by Winston Churchill ?We shall defend our island, whatever the cost may be, we shall fight on the beaches, we shall fight on the landing grounds, we shall fight in the fields and in the streets, we shall fight in the hills; we shall never surrender.? * Features of this release * Fix rendering failures * UI beautifications * Switch to themeable ruler * Speed optimisations * Show the project name in the window title * Requirements * gstreamer >= 0.10.24 * gst-python >= 0.10.16 * gnonlin >= 0.10.13 * pygtk >= 2.14.0 * Python >= 2.5 * zope.interface (http://www.zope.org/Products/ZopeInterface) * setuptools (http://peak.telecommunity.com/DevCenter/setuptools) * pygoocanvas (http://live.gnome.org/GooCanvas) * dbus and HAL for capture support * Contributors 39 Alessandro Decina 32 Brandon Lewis 22 Edward Hervey 2 Hendrik Richter 2 Jorge Gonz?lez 2 Michael Terry 1 Ant?n M?ixome 1 Claude Paroz 1 Daniel Nylander 1 Petr Kovar * Download PiTiVi source tarballs are available on gnome FTP: http://ftp.gnome.org/pub/GNOME/sources/pitivi/0.13/ See the website for distribution-specific packages. * Information and Feedback * Information for users and developers can be found on the PiTiVi website : http://www.pitivi.org/ * Comments and feedback are welcome. * Mailing-list : pitivi-pitivi at lists.sourceforge.net * PiTiVi bug-tracker : http://bugzilla.gnome.org/browse.cgi?product=pitivi * Bugs Fixed * 520653 : pitivi.desktop fixes and more * 575311 : clips appear to be duplicated at high zoom levels when the timeline cursor is near the edge * 576576 : not possible to add clip at end of timeline * 584170 : Video Playback Fails, Without Program Crash * 590114 : rendering never finishes * 590153 : Zoom to optimum level when loading projects * 590203 : image hang during playback (auriga) * 590440 : Strings not marked for translation in check.py * 591571 : Removing a keyframe is not easy * 591616 : crashes when creating a new project after loading a project * 591617 : regression: centering on playhead when zooming broke * 593736 : play button is empty * 594114 : Sound curves not taken into account when rendering * 594311 : Clicking the timeline leads to a TypeError * 575975 : project-centric window title * 582363 : huge tabs the size of a hallway * 594181 : playhead/seeking broken by latest changes to factory cache in singledecodebin * 594396 : " save as " confuses pitivi; subsequent saves are on the original project file From jaisena at yahoo.com Sat Sep 12 21:05:45 2009 From: jaisena at yahoo.com (jayasena s) Date: Sat, 12 Sep 2009 12:05:45 -0700 (PDT) Subject: [gst-devel] A/V sync and stutter issue In-Reply-To: <1252706520.13334.1.camel@fancy> Message-ID: <562003.1939.qm@web55303.mail.re4.yahoo.com> Hi Jan, Thanks for the suggestions. ? I would?start debugging the pipeline with gst-launch and totem. ? Also, Would like to know,?? When a buffer with decoded frame is sent using gst_pad_push() to downstream to?the xvimagesink , is there any event notification sent back with gst_pad_push() ?to the upstream elements, when the frame gets rendered at xvimagesink. If not with gst_pad_push(), is there any other API which can be used to send notification to the upstream element, after frame is rendered to the display from the xvimagesink. ? Thanks, Jayasena ? --- On Fri, 9/11/09, Jan Schmidt wrote: From: Jan Schmidt Subject: Re: [gst-devel] A/V sync and stutter issue To: "Discussion of the development of GStreamer" Date: Friday, September 11, 2009, 11:02 PM Hi, On Fri, 2009-09-11 at 11:22 +0100, Jan Schmidt wrote: Hrmn, my initial reply was lacking somewhat in detail ;) > On Thu, 2009-09-10 at 11:43 -0700, jayasena s wrote: > >? *snip* > >? ? ? ???Hi, > >? ? ? ? ? I am having A/V Sync and stutter issues , when using totem > >? ? ? ???player with gstreamer. > >? ? ? ???With gst-launch, audio and video are in sync and video is > >? ? ? ???smooth ( no stutter issue). > >? ? ? ? ? > >? ? ? ???Timestamps seem to be good, the stutter is more visible for > >? ? ? ???the first 4 seconds of the playback. > >? ? ? ? ? > >? ? ? ???Did anyone come across this issue, Could you suggest ways to > >? ? ? ???debug these issues I'd start by identifying the difference between the pipeline you're giving gst-launch and what totem is using. Totem (depending on the version) is using either playbin or playbin2, with the gconf audio and video sinks. Cheers, Jan. > >? ? ? ? ? > >? ? ? ???Thanks, > >? ? ? ???Jai > >? ? ? ??? > >? ? ? ??? > >? ? ? ??? > >? ? ? ???-----Inline Attachment Follows----- > >? ? ? ??? > >? ? ? ???------------------------------------------------------------------------------ > >? ? ? ???Let Crystal Reports handle the reporting - Free Crystal > >? ? ? ???Reports 2008 30-Day > >? ? ? ???trial. Simplify your report design, integration and deployment > >? ? ? ???- and focus on > >? ? ? ???what you do best, core application coding. Discover what's new > >? ? ? ???with > >? ? ? ???Crystal Reports now.? http://p.sf.net/sfu/bobj-july > >? ? ? ??? > >? ? ? ???-----Inline Attachment Follows----- > >? ? ? ??? > >? ? ? ???_______________________________________________ > >? ? ? ???gstreamer-devel mailing list > >? ? ? ???gstreamer-devel at lists.sourceforge.net > >? ? ? ???https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > >? ? ? ??? > > > > ------------------------------------------------------------------------------ > > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > > trial. Simplify your report design, integration and deployment - and focus on > > what you do best, core application coding. Discover what's new with > > Crystal Reports now.? http://p.sf.net/sfu/bobj-july > > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -- Jan Schmidt ------------------------------------------------------------------------------ Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day trial. Simplify your report design, integration and deployment - and focus on what you do best, core application coding. Discover what's new with Crystal Reports now.? http://p.sf.net/sfu/bobj-july _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: From azul at freaks-unidos.net Mon Sep 14 01:59:04 2009 From: azul at freaks-unidos.net (Alejandro Forero Cuervo) Date: Mon, 14 Sep 2009 01:59:04 +0200 Subject: [gst-devel] fork() and then use gstreamer functions? Message-ID: <20090913235904.GJ15987@freaks-unidos.net> I've been working in a music player. I figured it was time to get it to use gstreamer. I'm having some issues. The music player receives a path that it scans recursively for files. To do the scanning, it forks a child process that obtains the information about the files, one at a time, and reports the results back to the parent process over a pipe. Currently, it just uses readdir(2) and stat(2) and reports to the parent process the list of files it finds and their sizes. I've been modifying the code to hook gstreamer so that it can report metadata about the file. The problem I'm having is that when I call fork(2) and, after setting a pipeline, the child process calls gst_bus_poll, the child process seems to be trying to interact with the X server! Y get some assertion failures in xlib's code. I'm guessing gst_bus_poll just calls the glib primitives for the main loop, which still have the gtk-dependencies. I tried closing all files in the child (except the pipe used to talk to the parent and the standard error) but this just results in an assertion failure, Fatal IO error 9 (Bad file descriptor) on X server :0.0. So what's a programmer to do in order to fork a process and use gstreamer functions without having the child process attempt to keep track of all the widgets and stuff? Thanks! Alejo. http://azul.freaks-unidos.net/ From ds at entropywave.com Mon Sep 14 03:32:30 2009 From: ds at entropywave.com (David Schleef) Date: Sun, 13 Sep 2009 18:32:30 -0700 Subject: [gst-devel] fork() and then use gstreamer functions? In-Reply-To: <20090913235904.GJ15987@freaks-unidos.net> References: <20090913235904.GJ15987@freaks-unidos.net> Message-ID: <20090914013230.GA7368@entropywave.com> On Mon, Sep 14, 2009 at 01:59:04AM +0200, Alejandro Forero Cuervo wrote: ... > The problem I'm having is that when I call fork(2) and, after setting > a pipeline, the child process calls gst_bus_poll, the child process > seems to be trying to interact with the X server! Y get some > assertion failures in xlib's code. I'm guessing gst_bus_poll just > calls the glib primitives for the main loop, which still have the > gtk-dependencies. > > I tried closing all files in the child (except the pipe used to talk > to the parent and the standard error) but this just results in an > assertion failure, Fatal IO error 9 (Bad file descriptor) on X server > :0.0. > > So what's a programmer to do in order to fork a process and use > gstreamer functions without having the child process attempt to keep > track of all the widgets and stuff? You can't, at least in any useful manner. You could attempt to create a new GMainContext and main loop, and use those. That will prevent anything Gtk+ (or GStreamer) set up in the main loop of the primary process. Unfortunately, that's not the only shared resource. A reasonable scenario that cannot be worked around is that some thread in the main process holds a global lock (which are common in Glib and GStreamer) while another thread calls fork(). In the child process, this lock will never be unlocked, which can cause deadlocks. Another reasonable scenario is that the program could be running on an OS that doesn't support fork(), e.g., Windows. The fork() system call (when not immediately followed by exec()) is pretty useless in modern Unix programming. I'd recommend using a helper process, perhaps even the same executable with a --special-flag, or even just a separate thread. dave... From volter619 at 163.com Mon Sep 14 11:23:41 2009 From: volter619 at 163.com (Volter Yen) Date: Mon, 14 Sep 2009 17:23:41 +0800 (CST) Subject: [gst-devel] problem on freescale gstreamer plugin Message-ID: <17625784.515331252920221477.JavaMail.coremail@bj163app84.163.com> Hi all, I am working on freescale i.mx27 for mutltimedia application developing,according to the requirement I should construct a decode pipe and an encode pipe line at the same time. It seem to impossibe that using the freescale's decode and encode plugin at the same time. because they need to open the '/dev/vpu' when initialize the plugins for both decode plugin and decode plugin, but it is not permitted to do so, then the systemIOinit() could not be finished, that is only one pipe line is allowed. has anyone meet the similiar problem and how to work it out? please give me some hint on it. thank you! -------------- next part -------------- An HTML attachment was scrubbed... URL: From azul at freaks-unidos.net Mon Sep 14 11:45:39 2009 From: azul at freaks-unidos.net (Alejandro Forero Cuervo) Date: Mon, 14 Sep 2009 11:45:39 +0200 Subject: [gst-devel] fork() and then use gstreamer functions? In-Reply-To: <20090914013230.GA7368@entropywave.com> References: <20090913235904.GJ15987@freaks-unidos.net> <20090914013230.GA7368@entropywave.com> Message-ID: <20090914094539.GL15987@freaks-unidos.net> > > So what's a programmer to do in order to fork a process and use > > gstreamer functions without having the child process attempt to keep > > track of all the widgets and stuff? > > You can't, at least in any useful manner. You could attempt to > create a new GMainContext and main loop, and use those. That > will prevent anything Gtk+ (or GStreamer) set up in the main > loop of the primary process. Unfortunately, that's not the > only shared resource. > > A reasonable scenario that cannot be worked around is that some > thread in the main process holds a global lock (which are common > in Glib and GStreamer) while another thread calls fork(). In the > child process, this lock will never be unlocked, which can cause > deadlocks. I find it sad that, from your first ?reasonable scenario?, the fact that GStreamer uses threads prevents me from using fork and then the GStreamer functions. I was under the impression that I would be able to relatively ignore the heavily threaded nature of GStreamer from my application, as it's documentation has misleading statements such as ?an application does not need to be thread-aware in order to use GStreamer?. If I wanted to use GStreamer in my application, I would have to make it thread-aware to the point of not using fork(2) during normal operation, which translates to forcing me to using separate threads for certain things (ie. I need to scan directories or do DNS lookups). > Another reasonable scenario is that the program could be running > on an OS that doesn't support fork(), e.g., Windows. I don't care about those OSs for my application. :-) > The fork() system call (when not immediately followed by exec()) is > pretty useless in modern Unix programming. I'd recommend using > a helper process, perhaps even the same executable with a > --special-flag, or even just a separate thread. I disagree. I find it very useful. Of course, if you've accepted to use threads, sure, you then only need fork for a handful of cases. However, some of us prefer to avoid threads in general, ?and we don't mind the propagandistic application of the adjective ?modern? to ?Unix programming? to advocate the use of threads, we just counter it using the ?propagandistic? adverb, as you just saw. :-) fork is very useful for us. But, to each his own: I don't want to start a flame war about threads vs async-IO here. I will check to see how difficult it would be to use a helper process (which itself forks the processes that do the work and which communicate with the main process over named pipes). I wish I didn't have to add all this complexity to my program just to use GStreamer. :-( In other words, I hope you can find a way to, eventually, make it possible to use GStreamer from a forked son of a process that has also used it. Thanks for your help. :-) Alejo. http://azul.freaks-unidos.net/ From marc.leeman at gmail.com Mon Sep 14 12:00:57 2009 From: marc.leeman at gmail.com (Marc Leeman) Date: Mon, 14 Sep 2009 12:00:57 +0200 Subject: [gst-devel] problem on freescale gstreamer plugin In-Reply-To: <17625784.515331252920221477.JavaMail.coremail@bj163app84.163.com> References: <17625784.515331252920221477.JavaMail.coremail@bj163app84.163.com> Message-ID: <20090914100057.GE5472@crichton.homelinux.org> > I am working on freescale i.mx27 for mutltimedia application > developing,according to the requirement I should construct a decode pipe > and an encode pipe line at the same time. It seem to impossibe that > using the freescale's decode and encode plugin at the same time. because You are probably asking the wrong mailinglist: you are plugins by Freescale that are not availble in GStreamer. [mleeman at bane gst-git]$ gst-inspect encode No such element or plugin 'encode' [mleeman at bane gst-git]$ gst-inspect decode No such element or plugin 'decode' > they need to open the '/dev/vpu' when initialize the plugins for both > decode plugin and decode plugin, but it is not permitted to do so, > then the systemIOinit() could not be finished, that is only one pipe > line is allowed. [mleeman at bane 20090811]$ find . -name '*.[ch]' |xargs grep systemIOinit [mleeman at bane 20090811]$ find . -name '*.[ch]' |xargs grep vpu Both show nothing. > has anyone meet the similiar problem and how to work it out? please > give me some hint on it. thank you! You seem to be confusing GStreamer with custom hardware drivers from what I make out from your post. -- greetz, marc Those who can, do; those who can't, write. Those who can't write work for the Bell Labs Record. crichton 2.6.26 #1 PREEMPT Tue Jul 29 21:17:59 CDT 2008 GNU/Linux -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: Digital signature URL: From t.i.m at zen.co.uk Mon Sep 14 12:06:14 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Mon, 14 Sep 2009 11:06:14 +0100 Subject: [gst-devel] fork() and then use gstreamer functions? In-Reply-To: <20090913235904.GJ15987@freaks-unidos.net> References: <20090913235904.GJ15987@freaks-unidos.net> Message-ID: <1252922774.5022.6.camel@zingle> On Mon, 2009-09-14 at 01:59 +0200, Alejandro Forero Cuervo wrote: > I've been working in a music player. I figured it was time to get it > to use gstreamer. I'm having some issues. > (..) > The problem I'm having is that when I call fork(2) and, after setting > a pipeline, the child process calls gst_bus_poll, the child process > seems to be trying to interact with the X server! Y get some > assertion failures in xlib's code. I'm guessing gst_bus_poll just > calls the glib primitives for the main loop, which still have the > gtk-dependencies. > > I tried closing all files in the child (except the pipe used to talk > to the parent and the standard error) but this just results in an > assertion failure, Fatal IO error 9 (Bad file descriptor) on X server > :0.0. > > So what's a programmer to do in order to fork a process and use > gstreamer functions without having the child process attempt to keep > track of all the widgets and stuff? This sounds more like an issue with your GUI toolkit (Gtk+/Qt) than with GStreamer per se. GStreamer doesn't force you to use the GLib main loop - any GStreamer function that hooks into the main loop is just for convenience. In your case, you could use gst_bus_timed_pop_filtered() instead of gst_bus_poll(). That would avoid iterating the main loop. An external helper application with which you communicate via a socket or pipe or some other form of IPC might be a solution too (and makes debugging import issues a lot easier). Cheers -Tim From robert at ocallahan.org Sat Sep 12 00:18:17 2009 From: robert at ocallahan.org (Robert O'Callahan) Date: Sat, 12 Sep 2009 10:18:17 +1200 Subject: [gst-devel] [cairo] Using cairo/pixman for raw video in GStreamer In-Reply-To: References: Message-ID: <11e306600909111518v5f25b8f4i9e0a264697b84dfa@mail.gmail.com> Support for YUV formats in pixman/cairo would be useful for us in Gecko, and for other apps I'm sure. I'd love to see this in cairo! Rob -- "He was pierced for our transgressions, he was crushed for our iniquities; the punishment that brought us peace was upon him, and by his wounds we are healed. We all, like sheep, have gone astray, each of us has turned to his own way; and the LORD has laid on him the iniquity of us all." [Isaiah 53:5-6] -------------- next part -------------- An HTML attachment was scrubbed... URL: From k.kooi at student.utwente.nl Sat Sep 12 09:53:36 2009 From: k.kooi at student.utwente.nl (Koen Kooi) Date: Sat, 12 Sep 2009 09:53:36 +0200 Subject: [gst-devel] Using cairo/pixman for raw video in GStreamer In-Reply-To: References: Message-ID: On 11-09-09 19:02, Benjamin Otte wrote: > 1) "This is never gonna be fast enough" > I don't see why. Most of the operations people care about are just > memcpys and pixman is very good at detecting them and making them > fast. On the platforms I'm working on (ARM SoCs with 'video' hardware) everything that even resembles a memcpy is going to be slow. The effective DDR bandwidth is about 300MiB/s that is shared with the framebuffer. When using video there are a few things helping us: * Overlays that support YUV in hardware (with an XV driver) * Overlays that support scaling in hardware (with an XV driver) * NEON optimizations for various things in software And sometimes we even have a DSP to do the hard things for us (bitstream parsing, frame decoding), but even then you need to really take care not to do a memcpy, so you point the framebuffer pointer to the frame decoded by the DSP instead of copying it. I'm a high level kind of guy, so my question is: are such optimizations still possible with your proposal? regards, Koen From sandmann at daimi.au.dk Mon Sep 14 08:54:43 2009 From: sandmann at daimi.au.dk (Soeren Sandmann) Date: 14 Sep 2009 08:54:43 +0200 Subject: [gst-devel] [cairo] Using cairo/pixman for raw video in GStreamer In-Reply-To: References: Message-ID: Hi Benjamin, All of this sounds good to me. Below are a few comments on how YUV formats could be integrated in pixman. > 1) Add extensive YUV support to pixman Extensive YUV support would be a very useful addition to pixman. Apart from the benefits you listed, I think it also makes sense to have YUV support in XRender as a more powerful way of doing textured video than Xv. * Tiles Writing one pixel in a chroma subsampled format requires access to a 2x2 tile of RGB pixels, but the current general compositing only provides one scanline. A solution to that may be to move to a tiled architectured where general_composite() processes destination tiles instead of scanlines. This would require changing all the scanline accessors, but hopefully that is a mostly mechanical process. Aside from hopefully solving the subsampling problem, tiles would also have better cache behavior for rotated or filtered sources. * Format specification Pixman already has some support for YUV formats: PIXMAN_yuy2 = PIXMAN_FORMAT(16,PIXMAN_TYPE_YUY2,0,0,0,0), PIXMAN_yv12 = PIXMAN_FORMAT(12,PIXMAN_TYPE_YV12,0,0,0,0) but you can't write to them because of the subsampling problem mentioned above, and so there is pixman_format_supported_destination() API. It was probably a mistake to add that API, and future formats should always be supported for both reading and writing. Having a pixman_format_type_t like PIXMAN_TYPE_YUY2 and TYPE_YV12 for each video format is not going to scale, so we'll need some new scheme to describe video formats. I don't know enough about video formats to have an opinion on how to do this, but I don't think there is anything particularly great about the two exisiting format codes, so hopefully we can get away with deprecating them and respecifiying within the new scheme. Thanks, Soren From thomas at apestaart.org Mon Sep 14 12:45:14 2009 From: thomas at apestaart.org (Thomas Vander Stichele) Date: Mon, 14 Sep 2009 12:45:14 +0200 Subject: [gst-devel] 10 years of GStreamer Message-ID: <1252925114.3188.15.camel@otto.amantes> Marc-Andr? Lureau reminded me that this year we celebrate 10 years of GStreamer. Somehow we both thought the first release was done in October of 2009. Checking now, it seems that version 0.0.0 was actually released May 13 1999! So we've already past the ten year mark! Congratulations to all hackers past and present on an arguably successful Free Software project... Here's to another 10 years! Thomas -- She was a commited romantic and an anarcha-feminist. This was hard for her because it meant she couldn't blow up beautiful buildings. -- URGent, best radio on the net - 24/7 ! http://urgent.fm/ From bisht.sudarshan at gmail.com Mon Sep 14 13:24:27 2009 From: bisht.sudarshan at gmail.com (sudarshan bisht) Date: Mon, 14 Sep 2009 16:54:27 +0530 Subject: [gst-devel] 10 years of GStreamer In-Reply-To: <1252925114.3188.15.camel@otto.amantes> References: <1252925114.3188.15.camel@otto.amantes> Message-ID: <785339900909140424k49ae8a45tda9ecb53941922c0@mail.gmail.com> Congrates !!!!!!!!!!!!!!!!!!!!!! On Mon, Sep 14, 2009 at 4:15 PM, Thomas Vander Stichele < thomas at apestaart.org> wrote: > Marc-Andr? Lureau reminded me that this year we celebrate 10 years of > GStreamer. Somehow we both thought the first release was done in > October of 2009. > > Checking now, it seems that version 0.0.0 was actually released May 13 > 1999! > > So we've already past the ten year mark! Congratulations to all hackers > past and present on an arguably successful Free Software project... > > Here's to another 10 years! > > Thomas > > -- > She was a commited romantic and an anarcha-feminist. > This was hard for her > because it meant she couldn't blow up beautiful buildings. > -- > URGent, best radio on the net - 24/7 ! > http://urgent.fm/ > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus > on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Regards, Sudarshan Bisht -------------- next part -------------- An HTML attachment was scrubbed... URL: From otte at gnome.org Mon Sep 14 13:11:46 2009 From: otte at gnome.org (Benjamin Otte) Date: Mon, 14 Sep 2009 11:11:46 +0000 (UTC) Subject: [gst-devel] Using cairo/pixman for raw video in GStreamer References: <1252746698.6344.599.camel@odin.lan> Message-ID: Sebastian Dr?ge collabora.co.uk> writes: > As you can get a cairo_t from every surface, would this mean that all > cairo stuff has to support all the pixman surface formats? I mean, would > it be possible later to set YUV colors instead of RGB colors in cairo, > would it be possible to draw lines or whatever in cairo on some YUV > surface, etc? > In that case you probably have a lot of work to do ;) > Yes, the idea is that you have a function like this: cairo_surface_t *gst_cairo_create_surface (GstBuffer *buffer); It would look at the buffer's caps and create the right surface from it. So writing a colorspace element would look like this: static GstFlowReturn gst_cairo_colorspace_transform (GstBaseTransform * btrans, GstBuffer * inbuf, GstBuffer * outbuf) { cairo_surface_t *in, *out; cairo_t *cr; in = gst_cairo_create_surface (inbuf); out = gst_cairo_create_surface (outbuf); cr = cairo_create (out; cairo_set_source (cr, in); cairo_set_operator (cr, CAIRO_OPERATOR_SOURCE); cairo_paint (cr); cairo_destroy (cr); cairo_surface_destroy (in); cairo_surface_destroy (out); return GST_FLOW_OK; } Making that element also a video scaler is one more cairo_scale(). The work required to get this working is pretty small as both pixman and cairo are _very_ generic, you basically just need to implement a read_pixel and store_pixel vfunc for every format that returns an ARGB guint32 and everything will just work. The complexity comes from making it work fast, but even that is not hard, as pixman has a very sophisticated acceleration architecture. So it's just writing the accelerated versions, which I intend to do for all the ones that already exist and then focus on I420 and AYUV. > > 4) Port elements to use this cairo API > > Either add new elements (cairovideotestsrc, cairocolorspace) or add > > support for the old ones. While doing this, refine and improve cairo > > or pixman, so the elements can be implemented as nicely as possible. A > > lot of code inside GStreamer should go away > > That would be the same as gst-plugins-gl works nowadays. I'm all for it, > that's definitely a good idea. Not sure if we want a cairo dependency on > every video element now already though... > We could deprecate the old elements and add new ones or we could improvie the old ones. Not sure what the preferred way is. I guess historically gst has gone the "write new ones" route. Both ways should be equally feasible. > Which means that cairo/pixman must have a good framework in place to > also add new formats easily. If that's given it might make sense, yes. > > cairo/pixman should also support 8 bit and 16 bit grayscale and the > different Bayer formats too then btw. > As I said before: Adding new formats is easy, what is hard is making them fast. :) > Also it would mean, that if you have some codec that decodes into some > weird colorformat that is not supported by pixman/cairo yet, that you > need to wait for pixman/cairo to support it and gst-plugins-foo to > depend on that version or that you have to do conversions internally. > Well, there's quite a few formats that are only used by one or two codecs (like upside down raw video in AVI). I think it makes a lot of sense to not expose them and require separate elements for them. > All other steps might not make sense not sure what the cairo/pixman > people think about supporting random color formats. Also it would mean > that GStreamer depends on cairo as a required dependency. But I guess > cairo/pixman are at least portable enough to work everywhere. > Yeah, the hard dependency of plugins-base on cairo would be necessary. But considering cairo is a blessed dep today (textoverlay), it shouldn't be that hard to argue? Benjamin From ensonic at hora-obscura.de Mon Sep 14 13:49:23 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Mon, 14 Sep 2009 14:49:23 +0300 Subject: [gst-devel] Using cairo/pixman for raw video in GStreamer In-Reply-To: References: <1252746698.6344.599.camel@odin.lan> Message-ID: <4AAE2DC3.1070504@hora-obscura.de> Benjamin Otte schrieb: > Sebastian Dr?ge collabora.co.uk> writes: > > >> As you can get a cairo_t from every surface, would this mean that all >> cairo stuff has to support all the pixman surface formats? I mean, would >> it be possible later to set YUV colors instead of RGB colors in cairo, >> would it be possible to draw lines or whatever in cairo on some YUV >> surface, etc? >> In that case you probably have a lot of work to do ;) >> >> > Yes, the idea is that you have a function like this: > cairo_surface_t *gst_cairo_create_surface (GstBuffer *buffer); > It would look at the buffer's caps and create the right surface from it. > So writing a colorspace element would look like this: > static GstFlowReturn > gst_cairo_colorspace_transform (GstBaseTransform * btrans, GstBuffer * inbuf, > GstBuffer * outbuf) > { > cairo_surface_t *in, *out; > cairo_t *cr; > > in = gst_cairo_create_surface (inbuf); > out = gst_cairo_create_surface (outbuf); > cr = cairo_create (out; > cairo_set_source (cr, in); > cairo_set_operator (cr, CAIRO_OPERATOR_SOURCE); > cairo_paint (cr); > cairo_destroy (cr); > cairo_surface_destroy (in); > cairo_surface_destroy (out); > > return GST_FLOW_OK; > } > Making that element also a video scaler is one more cairo_scale(). > > The work required to get this working is pretty small as both pixman and cairo > are _very_ generic, you basically just need to implement a read_pixel and > store_pixel vfunc for every format that returns an ARGB guint32 and everything > will just work. > But having read_pixel/write_pixel will be slow. How can that be optimized (and/or vectorized). If I recall right "graphics-drivers" for turbo pascal in dos worked that way and they were slow! Stefan > The complexity comes from making it work fast, but even that is not hard, as > pixman has a very sophisticated acceleration architecture. So it's just writing > the accelerated versions, which I intend to do for all the ones that already > exist and then focus on I420 and AYUV. > > >>> 4) Port elements to use this cairo API >>> Either add new elements (cairovideotestsrc, cairocolorspace) or add >>> support for the old ones. While doing this, refine and improve cairo >>> or pixman, so the elements can be implemented as nicely as possible. A >>> lot of code inside GStreamer should go away >>> >> That would be the same as gst-plugins-gl works nowadays. I'm all for it, >> that's definitely a good idea. Not sure if we want a cairo dependency on >> every video element now already though... >> >> > We could deprecate the old elements and add new ones or we could improvie the > old ones. Not sure what the preferred way is. I guess historically gst has gone > the "write new ones" route. > Both ways should be equally feasible. > > >> Which means that cairo/pixman must have a good framework in place to >> also add new formats easily. If that's given it might make sense, yes. >> >> cairo/pixman should also support 8 bit and 16 bit grayscale and the >> different Bayer formats too then btw. >> >> > As I said before: Adding new formats is easy, what is hard is making them fast. > :) > > >> Also it would mean, that if you have some codec that decodes into some >> weird colorformat that is not supported by pixman/cairo yet, that you >> need to wait for pixman/cairo to support it and gst-plugins-foo to >> depend on that version or that you have to do conversions internally. >> >> > Well, there's quite a few formats that are only used by one or two codecs (like > upside down raw video in AVI). I think it makes a lot of sense to not expose > them and require separate elements for them. > > >> All other steps might not make sense not sure what the cairo/pixman >> people think about supporting random color formats. Also it would mean >> that GStreamer depends on cairo as a required dependency. But I guess >> cairo/pixman are at least portable enough to work everywhere. >> >> > Yeah, the hard dependency of plugins-base on cairo would be necessary. But > considering cairo is a blessed dep today (textoverlay), it shouldn't be that > hard to argue? > > Benjamin > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From katcipis at inf.ufsc.br Mon Sep 14 13:54:44 2009 From: katcipis at inf.ufsc.br (Tiago Katcipis) Date: Mon, 14 Sep 2009 08:54:44 -0300 Subject: [gst-devel] 10 years of GStreamer In-Reply-To: <785339900909140424k49ae8a45tda9ecb53941922c0@mail.gmail.com> References: <1252925114.3188.15.camel@otto.amantes> <785339900909140424k49ae8a45tda9ecb53941922c0@mail.gmail.com> Message-ID: <60a9403b0909140454u3c4e5989ueeb1036f103d98c0@mail.gmail.com> Congratulations to everyone involved on gstreamer development, it is a great framework, of course it is not perfect, but it is a very good example of how easy to use and well designed C code can be. Keep up the good job. On Mon, Sep 14, 2009 at 8:24 AM, sudarshan bisht wrote: > Congrates !!!!!!!!!!!!!!!!!!!!!! > > On Mon, Sep 14, 2009 at 4:15 PM, Thomas Vander Stichele < > thomas at apestaart.org> wrote: > >> Marc-Andr? Lureau reminded me that this year we celebrate 10 years of >> GStreamer. Somehow we both thought the first release was done in >> October of 2009. >> >> Checking now, it seems that version 0.0.0 was actually released May 13 >> 1999! >> >> So we've already past the ten year mark! Congratulations to all hackers >> past and present on an arguably successful Free Software project... >> >> Here's to another 10 years! >> >> Thomas >> >> -- >> She was a commited romantic and an anarcha-feminist. >> This was hard for her >> because it meant she couldn't blow up beautiful buildings. >> -- >> URGent, best radio on the net - 24/7 ! >> http://urgent.fm/ >> >> >> ------------------------------------------------------------------------------ >> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 >> 30-Day >> trial. Simplify your report design, integration and deployment - and focus >> on >> what you do best, core application coding. Discover what's new with >> Crystal Reports now. http://p.sf.net/sfu/bobj-july >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > > > -- > Regards, > > Sudarshan Bisht > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus > on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em menos de 5 minutos aparecer? algu?m pra discordar de sua escolha de distribui??o, do particionamento, do gerenciador de janelas, do ambiente de desktop, do editor de textos..." -------------- next part -------------- An HTML attachment was scrubbed... URL: From otte at gnome.org Mon Sep 14 15:57:38 2009 From: otte at gnome.org (Benjamin Otte) Date: Mon, 14 Sep 2009 13:57:38 +0000 (UTC) Subject: [gst-devel] Using cairo/pixman for raw video in GStreamer References: <1252746698.6344.599.camel@odin.lan> <4AAE2DC3.1070504@hora-obscura.de> Message-ID: Stefan Kost hora-obscura.de> writes: > But having read_pixel/write_pixel will be slow. How can that be > optimized (and/or vectorized). If I recall right "graphics-drivers" for > turbo pascal in dos worked that way and they were slow! > As I said, pixman is very sophisticated. The first step is implementing the pixman_image ops - see http://cgit.freedesktop.org/pixman/tree/pixman/pixman-private.h#n160 - I think fetch_scanline_raw_32 and store_scanline_raw_32 are required, the rest can will use defaults. Once you've done that, all cairo ops work on this format. And it'll probably be incredibly slow. After that, you have a look at pixman_implementation_t - see http://cgit.freedesktop.org/pixman/tree/pixman/pixman-private.h#n372 - and realize that for every rendering operation, there is a vfunc that you can implement specifically tuned for that operation. So you can make "AYUV OVER I420" or "I420 SOURCE RGB" really fast and keep "YVU9 DIFFERENCE NV12" as slow as you want. Of course, that's a lot of work. So there's also steps in between, like the ability to implement blit and fill function for simple copies or fills with a single color - see http://cgit.freedesktop.org/pixman/tree/pixman/pixman-private.h#n405 - which get called automatically by the general functions. Of course, those functions can be separately optimized for different architectures - see http://cgit.freedesktop.org/pixman/tree/pixman/pixman-arm-neon.c for an example - so you can do all the optimizations you want. So I fail to see any reason why some important code would be slow. Benjamin From ensonic at hora-obscura.de Mon Sep 14 17:03:04 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Mon, 14 Sep 2009 18:03:04 +0300 Subject: [gst-devel] Using cairo/pixman for raw video in GStreamer In-Reply-To: References: <1252746698.6344.599.camel@odin.lan> <4AAE2DC3.1070504@hora-obscura.de> Message-ID: <4AAE5B28.9040600@hora-obscura.de> Benjamin Otte schrieb: > Stefan Kost hora-obscura.de> writes: > > >> But having read_pixel/write_pixel will be slow. How can that be >> optimized (and/or vectorized). If I recall right "graphics-drivers" for >> turbo pascal in dos worked that way and they were slow! >> >> > As I said, pixman is very sophisticated. > > The first step is implementing the pixman_image ops - see > http://cgit.freedesktop.org/pixman/tree/pixman/pixman-private.h#n160 - I think > fetch_scanline_raw_32 and store_scanline_raw_32 are required, the rest can will > use defaults. > Once you've done that, all cairo ops work on this format. And it'll probably be > incredibly slow. > > After that, you have a look at pixman_implementation_t - see > http://cgit.freedesktop.org/pixman/tree/pixman/pixman-private.h#n372 - and > realize that for every rendering operation, there is a vfunc that you can > implement specifically tuned for that operation. So you can make "AYUV OVER > I420" or "I420 SOURCE RGB" really fast and keep "YVU9 DIFFERENCE NV12" as slow > as you want. > > Of course, that's a lot of work. So there's also steps in between, like the > ability to implement blit and fill function for simple copies or fills with a > single color - see > http://cgit.freedesktop.org/pixman/tree/pixman/pixman-private.h#n405 - which get > called automatically by the general functions. > > Of course, those functions can be separately optimized for different > architectures - see > http://cgit.freedesktop.org/pixman/tree/pixman/pixman-arm-neon.c for an example > - so you can do all the optimizations you want. > > So I fail to see any reason why some important code would be slow. > That sounds better then, but also mean that the work is not pretty small as you've said earlier. Anyway that is to be expected. Now we just need something like orc that can help you to write all those variants for all those datatypes :) Stefan > Benjamin > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From jcastellanos77 at gmail.com Mon Sep 14 18:19:57 2009 From: jcastellanos77 at gmail.com (Joaquin Castellanos) Date: Mon, 14 Sep 2009 11:19:57 -0500 Subject: [gst-devel] Info for codec_data parsing for H264 in a FLV container In-Reply-To: <4AA67764.3090302@hora-obscura.de> References: <400edb7e0909041031q40f71641t8b4b738b85d465ec@mail.gmail.com> <4AA67764.3090302@hora-obscura.de> Message-ID: <400edb7e0909140919m10bfe54k1c550bb123be8e57@mail.gmail.com> Thanks! On Tue, Sep 8, 2009 at 10:25 AM, Stefan Kost wrote: > Joaquin Castellanos schrieb: > > Hi > > > > I am looking for information required to parse H264 codec_data. > > With some Flv containers (with H264 v-streams) the flvdemux does not > > parse width, height or framerate, > > instead the demuxer sent the codec_data to the next element. > > > > e.g. > > > > # gst-launch filesrc location > > =/data/EVM_filesystems/x0089714/target/Vid.flv ! flvdemux name=demux > > demux.video ! fakesink -v > > Setting pipeline to PAUSED ... > > /GstPipeline:pipeline0/GstFLVDemux:demux.GstPad:video: caps = > > video/x-h264, pixel-aspect-ratio=(fraction)1/1, > > > codec_data=(buffer)01640033ffe1001c67640033ac2cc502d0ceffc01400144400000fa40003a9823c60c65801000468eebcb0 > > > > > have a look at h264parse. > > Stefan > > > Best regards > > Joaquin > > ------------------------------------------------------------------------ > > > > > ------------------------------------------------------------------------------ > > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 > 30-Day > > trial. Simplify your report design, integration and deployment - and > focus on > > what you do best, core application coding. Discover what's new with > > Crystal Reports now. http://p.sf.net/sfu/bobj-july > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > trial. Simplify your report design, integration and deployment - and focus > on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From shawn_mcmurdo at yahoo.com Tue Sep 15 02:19:13 2009 From: shawn_mcmurdo at yahoo.com (Shawn McMurdo) Date: Mon, 14 Sep 2009 17:19:13 -0700 (PDT) Subject: [gst-devel] Why audio extract on flv won't preroll? Message-ID: <818311.52191.qm@web53608.mail.re2.yahoo.com> Hi, I am trying to do an audio extract on various types of videos. Using decodebin as shown below on this video: http://www.shawnim.com/ax/balboa.flv causes gstreamer to hang in the preroll. The same pipeline works on other flvs and wmvs. Can anyone explain why this is happening? I am running gstreamer 0.10.24 on Fedora 11 (2.6.29.6-217.2.16.fc11.x86_64). The pipeline I am using is: gst-launch-0.10 -m -t -v filesrc location=$1 ! decodebin ! audioconvert ! audio/x-raw-int,channels=1,width=16,depth=16 ! audioresample ! audio/x-raw-int,rate=16000 ! wavenc ! filesink location=$1.wav The output from running this pipeline is: http://www.shawnim.com/ax/balboa.out BTW, decodebin2 works on this file but seems to have trouble with other files. Any explanations or pointer appreciated. Thanks. Shawn -------------- next part -------------- An HTML attachment was scrubbed... URL: From volter619 at 163.com Tue Sep 15 03:35:30 2009 From: volter619 at 163.com (Volter Yen) Date: Tue, 15 Sep 2009 09:35:30 +0800 (CST) Subject: [gst-devel] problem on freescale gstreamer plugin In-Reply-To: <20090914100057.GE5472@crichton.homelinux.org> References: <20090914100057.GE5472@crichton.homelinux.org> <17625784.515331252920221477.JavaMail.coremail@bj163app84.163.com> Message-ID: <16712171.57671252978530367.JavaMail.coremail@bj163app21.163.com> Hi marc, I know this mailist is mainly focused on the opensource gstreamer items, but I think there are still many people on ti or freescale's hardware platform using partly the gstreamer opensource framework. and then they would be in this maillist too. when they notice this problem, they expirenced engineer would give me some hint on this issue. Anyway, thank you for the reply. ?2009-09-14?"Marc Leeman" ??? >> I am working on freescale i.mx27 for mutltimedia application >> developing,according to the requirement I should construct a decode pipe >> and an encode pipe line at the same time. It seem to impossibe that >> using the freescale's decode and encode plugin at the same time. because > >You are probably asking the wrong mailinglist: you are plugins by >Freescale that are not availble in GStreamer. > >[mleeman at bane gst-git]$ gst-inspect encode >No such element or plugin 'encode' >[mleeman at bane gst-git]$ gst-inspect decode >No such element or plugin 'decode' > >> they need to open the '/dev/vpu' when initialize the plugins for both >> decode plugin and decode plugin, but it is not permitted to do so, >> then the systemIOinit() could not be finished, that is only one pipe >> line is allowed. > >[mleeman at bane 20090811]$ find . -name '*.[ch]' |xargs grep systemIOinit >[mleeman at bane 20090811]$ find . -name '*.[ch]' |xargs grep vpu > >Both show nothing. > >> has anyone meet the similiar problem and how to work it out? please >> give me some hint on it. thank you! > >You seem to be confusing GStreamer with custom hardware drivers from >what I make out from your post. > > >-- > greetz, marc >Those who can, do; those who can't, write. >Those who can't write work for the Bell Labs Record. >crichton 2.6.26 #1 PREEMPT Tue Jul 29 21:17:59 CDT 2008 GNU/Linux -------------- next part -------------- An HTML attachment was scrubbed... URL: From Florian.Feil at eads.com Tue Sep 15 17:00:38 2009 From: Florian.Feil at eads.com (Feil, Florian) Date: Tue, 15 Sep 2009 17:00:38 +0200 Subject: [gst-devel] UDPSRC BUG Message-ID: <881487A9950B044AA6861B700374AAAF02C3C013@deimsg40.de.net.world> Hallo, I would like to play a multicast Stream from a Camera. It streams a 4cif codec. Under Linux it works fine but i neet to play the Stream under Windows. I would open the stream with the following command: gst-launch-0.10.exe udpsrc uri=udp://225.2.47.168:60000, caps=\"application/x-rtp\" ! rtph263depay ! ffdec_h263 ! Directdrawsink Here ist the output: C:\Users\fefl100\workspace\svn\Build\Windows\Win32\Release\bin\gstreamer>gst-launch-0.10.exe udpsrc uri=udp://225.2.47.168:60000, caps=\"application/x-rtp\" ! rtph263depay ! ffdec_h263 ! directdrawsink Setting pipeline to PAUSED ... ERROR: Pipeline doesn't want to pause. ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Could not get/set settings from/on resource. Additional debug info: ..\..\..\Source\gst-plugins-good\gst\udp\gstudpsrc.c(940): gst_udpsrc_start (): /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: bind failed -1: No error (0) Setting pipeline to NULL ... Freeing pipeline ... Then I build the newest Trunk Version form the SVN. The result is the same. Thanks for help Mit freundlichen Gr??en - Kind regards Florian Feil -------------- next part -------------- An HTML attachment was scrubbed... URL: From spitzak at gmail.com Mon Sep 14 20:00:19 2009 From: spitzak at gmail.com (Bill Spitzak) Date: Mon, 14 Sep 2009 11:00:19 -0700 Subject: [gst-devel] [cairo] Using cairo/pixman for raw video in GStreamer In-Reply-To: References: Message-ID: <4AAE84B3.7070203@gmail.com> Soeren Sandmann wrote: > Hi Benjamin, > > All of this sounds good to me. Below are a few comments on how YUV > formats could be integrated in pixman. > >> 1) Add extensive YUV support to pixman > > Extensive YUV support would be a very useful addition to pixman. Apart > from the benefits you listed, I think it also makes sense to have YUV > support in XRender as a more powerful way of doing textured video than > Xv. > > * Tiles > > Writing one pixel in a chroma subsampled format requires access to a > 2x2 tile of RGB pixels, but the current general compositing only > provides one scanline. This is true of 4:1:1 (and other things with "1" in them but that is the only common one). Most of my work with compressed YUV has been with 4:2:2 which is alternating yuyvyuyv... and can be directly written from a single scanline. > A solution to that may be to move to a tiled architectured where > general_composite() processes destination tiles instead of > scanlines. This would require changing all the scanline accessors, but > hopefully that is a mostly mechanical process. This probably won't help if the borders of the tiles do not line up with the blocks needed. If the input is translated by 1 pixel vertically then you will need multiple tiles to write portions of the image. In general I consider tiled apis to make things unnecessarily complicated. The majority of cairo input is packed into an array. You either need to require images to be padded out to a multiple of tile size, or you need to greatly complicate things with "partial tiles" with whatever code is needed to avoid ever addressing the non-existent parts of the tiles. Scanlines are instead trivial to extract from a packed array, and save the overhead of having to think about the vertical iterator, allow translation and cropping and vertical flipping to be done in-place, and require small enough amounts of temporary storage that it tends to be done in the cpu cache. I think storage into 4:1:1 must be done by keeping the previous scanline in a temporary buffer and combining them in the scanline processor that is writing to the buffer. > Aside from hopefully solving the subsampling problem, tiles would also > have better cache behavior for rotated or filtered sources. No the performance is TERRIBLE for filters. A filter near the edge of a tile will require an entire neighboring tile. In scanlines the filter always gets only the exact input scanlines needed. Tiles do help for rotation of giant images, and for drawing a section out of the center of an image. But neither of these are common operations for Cairo, which really wants to draw images that are smaller than the screen fast. Tiles are also helpful for operations that only change a small portion of the image. But I doubt cairo is going to be altered to use a referenced-counted set of tiles for all storage, I suspect it is much faster to always use memory laid out such that it is in the form that the hardware wants. This makes it impossible to use this advantage of tiles. I think compression of YUV data would have to be done by buffers on I/O. Except for detecting literal copies with a translation that allows it, I don't think cairo or pixman should attempt to do anything with compressed YUV, just like it does not attempt to do anything with compressed jpeg data. It should decompose it into YUV channels. Even then YUV support requires special code as it is not RGB with different primaries. Black is .5 in the UV channels. If this is to be fast at all all the compositing operations need to be changed. It is not as bad as it might look, the interdependence of the cahnnels should cancel out of the math, but the calculation of UV will be differnt than the ones for Y and RGB. From Florian.Feil at eads.com Tue Sep 15 16:57:05 2009 From: Florian.Feil at eads.com (Feil, Florian) Date: Tue, 15 Sep 2009 16:57:05 +0200 Subject: [gst-devel] UDPSRC BUG Message-ID: <881487A9950B044AA6861B700374AAAF02C3C012@deimsg40.de.net.world> Hallo, I would like to play a multicast Stream from a Camera. It streams a 4cif codec. Under Linux it works fine but i neet to play the Stream under Windows. I would open the stream with the following command: gst-launch-0.10.exe udpsrc uri=udp://225.2.47.168:60000, caps=\"application/x-rtp\" ! rtph263depay ! ffdec_h263 ! Directdrawsink Here ist the output: C:\Users\fefl100\workspace\svn\Build\Windows\Win32\Release\bin\gstreamer>gst-launch-0.10.exe udpsrc uri=udp://225.2.47.168:60000, caps=\"application/x-rtp\" ! rtph263depay ! ffdec_h263 ! directdrawsink Setting pipeline to PAUSED ... ERROR: Pipeline doesn't want to pause. ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Could not get/set settings from/on resource. Additional debug info: ..\..\..\Source\gst-plugins-good\gst\udp\gstudpsrc.c(940): gst_udpsrc_start (): /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: bind failed -1: No error (0) Setting pipeline to NULL ... Freeing pipeline ... Then I build the newest Trunk Version form the SVN. The result is the same. Thanks for help Mit freundlichen Gr??en - Kind regards Florian Feil -------------- next part -------------- An HTML attachment was scrubbed... URL: From julien.isorce at gmail.com Tue Sep 15 17:52:12 2009 From: julien.isorce at gmail.com (Julien Isorce) Date: Tue, 15 Sep 2009 17:52:12 +0200 Subject: [gst-devel] UDPSRC BUG In-Reply-To: <881487A9950B044AA6861B700374AAAF02C3C012@deimsg40.de.net.world> References: <881487A9950B044AA6861B700374AAAF02C3C012@deimsg40.de.net.world> Message-ID: <180a127d0909150852mca47c9ela0b1325589aa512b@mail.gmail.com> Hi, Does this pipeline works: gst-launch-0.10.exe udpsrc port=200 ! fakesink ? If not, try to compile the udp plugin with mingw >>Then I build the newest Trunk Version form the SVN. The result is the same. if you are currently using VS. (with the current git, udpsink fails with this error (and compiled with mingw) WARN multiudpsink gstmultiudpsink.c:790:gst_multiudpsink_init_send: error: Could not set TTL socket option (0): No error ERROR: Pipeline doesn't want to pause. ERROR: from element /GstPipeline:pipeline0/GstUDPSink:udpsink0: Could not get/set settings from/on resource. Additional debug info: gstmultiudpsink.c(790): gst_multiudpsink_init_send (): /GstPipeline:pipeline0/GstUDPSink:udpsink0: Could not set TTL socket option (0): No error Setting pipeline to NULL ... ) Julien 2009/9/15 Feil, Florian > Hallo, > > I would like to play a multicast Stream from a Camera. It streams a 4cif > codec. Under Linux it works fine but i neet to play the Stream under > Windows. I would open the stream with the following command: > > gst-launch-0.10.exe udpsrc uri=udp://225.2.47.168:60000, > caps=\"application/x-rtp\" ! rtph263depay ! ffdec_h263 ! Directdrawsink > > Here ist the output: > > C:\Users\fefl100\workspace\svn\Build\Windows\Win32\Release\bin\gstreamer>gst-launch-0.10.exe > udpsrc uri=udp://225.2.47.168:60000, caps=\"application/x-rtp\" ! > rtph263depay ! ffdec_h263 ! directdrawsink > > Setting pipeline to PAUSED ... > ERROR: Pipeline doesn't want to pause. > ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Could not > get/set settings from/on resource. > Additional debug info: > ..\..\..\Source\gst-plugins-good\gst\udp\gstudpsrc.c(940): gst_udpsrc_start > (): /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: > > bind failed -1: No error (0) > Setting pipeline to NULL ... > Freeing pipeline ? > > Then I build the newest Trunk Version form the SVN. The result is the same. > > Thanks for help > > Mit freundlichen Gr??en - Kind regards > > Florian Feil > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From fabioestevam at yahoo.com Tue Sep 15 18:37:30 2009 From: fabioestevam at yahoo.com (Fabio Estevam) Date: Tue, 15 Sep 2009 09:37:30 -0700 (PDT) Subject: [gst-devel] problem on freescale gstreamer plugin In-Reply-To: <17625784.515331252920221477.JavaMail.coremail@bj163app84.163.com> Message-ID: <793343.74778.qm@web51011.mail.re2.yahoo.com> Volter, You should enter a service request at www.freescale.com/support Regards, Fabio Estevam --- On Mon, 9/14/09, Volter Yen wrote: > From: Volter Yen > Subject: [gst-devel] problem on freescale gstreamer plugin > To: "gstreamer-devel at lists.sourceforge.net" > Date: Monday, September 14, 2009, 6:23 AM > Hi all, > ??? I am working on freescale i.mx27 for > mutltimedia application developing,according to?the > requirement I should construct a decode pipe and an encode > pipe line at the same time.?? ?It seem to > impossibe that using the freescale's decode and encode > plugin at the same time. because?they need to open the > '/dev/vpu'? when initialize the plugins > for?both decode plugin and decode plugin,? but it > is not permitted to do so,?then the?systemIOinit() > could not?be finished, that is only one pipe line is > allowed.? > ???? has?anyone meet? the > similiar problem and how to work it out? please give me some > hint on?it.? thank you!? > ? > > > "????",????60??? > > -----Inline Attachment Follows----- > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal > Reports 2008 30-Day > trial. Simplify your report design, integration and > deployment - and focus on > what you do best, core application coding. Discover what's > new with > Crystal Reports now.? http://p.sf.net/sfu/bobj-july > -----Inline Attachment Follows----- > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From msmith at xiph.org Tue Sep 15 18:59:25 2009 From: msmith at xiph.org (Michael Smith) Date: Tue, 15 Sep 2009 09:59:25 -0700 Subject: [gst-devel] UDPSRC BUG In-Reply-To: <180a127d0909150852mca47c9ela0b1325589aa512b@mail.gmail.com> References: <881487A9950B044AA6861B700374AAAF02C3C012@deimsg40.de.net.world> <180a127d0909150852mca47c9ela0b1325589aa512b@mail.gmail.com> Message-ID: <3c1737210909150959y2d51346bibf060169902dc9c@mail.gmail.com> On Tue, Sep 15, 2009 at 8:52 AM, Julien Isorce wrote: > Hi, > > Does this pipeline works: gst-launch-0.10.exe udpsrc port=200 ! fakesink > ? > > If not, try to compile the udp plugin with mingw > >>>Then I build the newest Trunk Version form the SVN. The result is the >>> same. > > if you are currently using VS. > > (with the current git, udpsink fails with this error (and compiled with > mingw) > WARN????????? multiudpsink > gstmultiudpsink.c:790:gst_multiudpsink_init_send: error: Could not > set TTL socket option (0): No error I believe this patch makes multiudpsink work on windows: https://bugzilla.gnome.org/show_bug.cgi?id=534243 Note that I'm still waiting for more windows testing (particularly on vista), and hopefully a code review from someone who knows this stuff. udpsrc might also have similar issues, I'm not sure - I can fix these if people are willing to test the code and comment on that bug. Mike From startoftext at gmail.com Tue Sep 15 19:43:39 2009 From: startoftext at gmail.com (startoftext) Date: Tue, 15 Sep 2009 10:43:39 -0700 (PDT) Subject: [gst-devel] alsasrc device=hw:1 pipeline problems Message-ID: <25458576.post@talk.nabble.com> Can any one here tell me why this pipeline does not work? gst-launch-0.10 v4l2src ! queue ! ffmpegcolorspace ! ffenc_flv ! queue ! ffmux_flv name=mux alsasrc device=hw:1 ! queue ! audioconvert ! lame ! queue ! mux. mux. ! filesink location=vid.flv Here is the output i get from above: Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstAudioSrcClock ERROR: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Internal data flow error. Additional debug info: gstbasesrc.c(2334): gst_base_src_loop (): /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: streaming task paused, reason not-negotiated (-4) Execution ended after 37319285 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... Setting pipeline to NULL ... Freeing pipeline ... Funny thing is that if i change the device from hw:1 to hw:0 it works. Also this pipeline below DOES work. gst-launch-0.10 v4l2src ! queue ! xvimagesink . alsasrc device=hw:1 ! alsasink Any one have any ideas? -- View this message in context: http://www.nabble.com/alsasrc-device%3Dhw%3A1-pipeline-problems-tp25458576p25458576.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From gerecke at gmail.com Tue Sep 15 23:54:37 2009 From: gerecke at gmail.com (attaboy) Date: Tue, 15 Sep 2009 14:54:37 -0700 (PDT) Subject: [gst-devel] UDPSRC BUG Message-ID: <25461052.post@talk.nabble.com> I can confirm that udpsrc is broken on Windows and I'm willing to test fixes. I've just spent more time than I'd like to admit trying to get a multi-cast receive working on Windows XP using this simple pipeline: gst-launch udpsrc uri="udp://224.1.2.3:1234" ! fakesink I tried the 0.10.15 udpsrc plugin with this patch (https://forja.rediris.es/forum/message.php?msg_id=180551) and had no success with failure on the bind. I backed up to 0.10.8 (before IP6 support) and still had failure on the bind. I hard coded the local address to INADDR_ANY right before the bind and success! /* if (src->multi_addr.imr_multiaddr.s_addr) src->myaddr.sin_addr.s_addr = src->multi_addr.imr_multiaddr.s_addr; else */ src->myaddr.sin_addr.s_addr = INADDR_ANY; I tried hardwiring INADDR_ANY with the patched 0.10.15 version and ran with no crashes, but also received no data, so right now I'm back to 0.10.8. Point me to some code/patches and I'll test on W2K3, XP and Windows 7. I'm building using the OSS build system as opposed to mingw, hopefully that's not a problem. Bill Michael Smith-59 wrote: > > On Tue, Sep 15, 2009 at 8:52 AM, Julien Isorce > wrote: >> Hi, >> >> Does this pipeline works: gst-launch-0.10.exe udpsrc port=200 ! fakesink >> ? >> >> If not, try to compile the udp plugin with mingw >> >>>>Then I build the newest Trunk Version form the SVN. The result is the >>>> same. >> >> if you are currently using VS. >> >> (with the current git, udpsink fails with this error (and compiled with >> mingw) >> WARN????????? multiudpsink >> gstmultiudpsink.c:790:gst_multiudpsink_init_send: error: Could >> not >> set TTL socket option (0): No error > > I believe this patch makes multiudpsink work on windows: > https://bugzilla.gnome.org/show_bug.cgi?id=534243 > > Note that I'm still waiting for more windows testing (particularly on > vista), and hopefully a code review from someone who knows this stuff. > > udpsrc might also have similar issues, I'm not sure - I can fix these > if people are willing to test the code and comment on that bug. > > Mike > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register > now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- View this message in context: http://www.nabble.com/UDPSRC-BUG-tp25455747p25461052.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From ping.gao at harman.com Wed Sep 16 01:44:12 2009 From: ping.gao at harman.com (Gao, Ping) Date: Tue, 15 Sep 2009 18:44:12 -0500 Subject: [gst-devel] What is the proper pipeline for YouTube video playback? Message-ID: <6DF262F0BAB5944AB03D1FE91354A1EF05FFE148AE@HICGWSEX01.ad.harman.com> Hi all: Has anybody developed pipeline to play YouTube video (MPEG4 video and AAC audio) through gstreamer rtspsrc, etc.? I am running on FC9 with the following packages installed: gstreamer-0.10.24 gst-plugins-base-0.10.24 gst-plugins-good-0.10.8 gst-plugins-bad-0.10.14 gst-plugins-ugly-0.10.12 gst-ffmpeg-0.10.8 Here is the command line I used and the error messages: [root at localhost zgrviewer]# gst-launch rtspsrc location="rtsp://rtsp2.youtube.com/CiQLENy73wIaGwnYRKJ3bPTBdBMYESARFEgGUghzdGFuZGFyZAw=/0/0/0/video.3gp" ! rtph264depay ! ffdec_h264 ! xvimagesinkSetting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstSystemClock ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc2: Internal data flow error. Additional debug info: gstbasesrc.c(2378): gst_base_src_loop (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc2: streaming task paused, reason not-linked (-1) Execution ended after 2758843655 ns. Setting pipeline to PAUSED ... Setting pipeline to READY ... Setting pipeline to NULL ... Freeing pipeline ... What is missing in my command? If anybody has some sample application code for YOuTuve video playback, that will help too. Thanks Ping -------------- next part -------------- An HTML attachment was scrubbed... URL: From sebastian.droege at collabora.co.uk Wed Sep 16 09:17:01 2009 From: sebastian.droege at collabora.co.uk (Sebastian =?ISO-8859-1?Q?Dr=F6ge?=) Date: Wed, 16 Sep 2009 09:17:01 +0200 Subject: [gst-devel] RELEASE: GStreamer C# bindings 0.9.1 "(Almost) nothing new here" Message-ID: <1253085421.4677.15.camel@odin.lan> This mail announces the release of the GStreamer C# bindings 0.9.1 "(Almost) nothing new here". The GStreamer C# Bindings are bindings for the GStreamer 0.10 release series and selected libraries and plugins. It comes with a number of examples. For more information, see http://gstreamer.freedesktop.org/modules/gstreamer-sharp.html To file bugs, go to http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gst-sharp Enjoy! PS: Note that this release doesn't make any API/ABI stability guarantees, see the release notes and the README for more information! -------------- next part -------------- Release notes for GStreamer C# bindings?0.9.1 "(Almost) nothing new here" The GStreamer team is proud to announce a new development release of the GStreamer C# bindings for the GStreamer 0.10.x release series. The GStreamer 0.10.x release series is a stable series targeted at end users. Please note that at this time the GStreamer C# bindings are not consindered API/ABI stable, and public interfaces may still change from release to release. These changes will most likely be small. Please read the README file for more information on this. Features of this release * Update bindings to GStreamer 0.10.24 * Update internal glib-sharp copy to latest SVN trunk * Some cleanup Bugs fixed in this release * 594127 : gstreamer-sharp does not build Download You can find source releases of gstreamer-sharp in the download directory: http://gstreamer.freedesktop.org/src/gstreamer-sharp/ GStreamer Homepage More details can be found on the project's website: http://gstreamer.freedesktop.org/ Support and Bugs We use GNOME's bugzilla for bug reports and feature requests: http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer Developers GStreamer is stored in Git, hosted at git.freedesktop.org, and can be cloned from there. Interested developers of the core library, plug-ins, and applications should subscribe to the gstreamer-devel list. If there is sufficient interest we will create more lists as necessary. Applications Contributors to this release * Gabriel Burt * Sebastian Dr?ge ? -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Dies ist ein digital signierter Nachrichtenteil URL: From ensonic at hora-obscura.de Wed Sep 16 10:45:08 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Wed, 16 Sep 2009 11:45:08 +0300 Subject: [gst-devel] alsasrc device=hw:1 pipeline problems In-Reply-To: <25458576.post@talk.nabble.com> References: <25458576.post@talk.nabble.com> Message-ID: <4AB0A594.80007@hora-obscura.de> startoftext schrieb: > Can any one here tell me why this pipeline does not work? > > gst-launch-0.10 v4l2src ! queue ! ffmpegcolorspace ! ffenc_flv ! queue ! > ffmux_flv name=mux alsasrc device=hw:1 ! queue ! audioconvert ! lame ! queue > ! mux. mux. ! filesink location=vid.flv > > Here is the output i get from above: > Setting pipeline to PAUSED ... > Pipeline is live and does not need PREROLL ... > Setting pipeline to PLAYING ... > New clock: GstAudioSrcClock > ERROR: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Internal > data flow error. > Additional debug info: > gstbasesrc.c(2334): gst_base_src_loop (): > /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: > streaming task paused, reason not-negotiated (-4) > Execution ended after 37319285 ns. > Setting pipeline to PAUSED ... > Setting pipeline to READY ... > Setting pipeline to NULL ... > Freeing pipeline ... > > > Funny thing is that if i change the device from hw:1 to hw:0 it works. Also > this pipeline below DOES work. > > gst-launch-0.10 v4l2src ! queue ! xvimagesink . alsasrc device=hw:1 ! > alsasink > what you need to get working is alsasrc device=hw:1 ! queue ! audioconvert ! lame ! fakesink I suspect sampling rates, so please try alsasrc device=hw:1 ! queue ! audioresample ! audioconvert ! lame ! fakesink Also consider to use less queues. Does that work? gst-launch-0.10 v4l2src ! ffmpegcolorspace ! ffenc_flv ! queue ! ffmux_flv name=mux alsasrc device=hw:1 ! audioresample ! audioconvert ! lame ! queue ! mux. mux. ! filesink location=vid.flv Stefan > Any one have any ideas? > From bilboed at gmail.com Wed Sep 16 13:05:50 2009 From: bilboed at gmail.com (Edward Hervey) Date: Wed, 16 Sep 2009 13:05:50 +0200 Subject: [gst-devel] What is the proper pipeline for YouTube video playback? In-Reply-To: <6DF262F0BAB5944AB03D1FE91354A1EF05FFE148AE@HICGWSEX01.ad.harman.com> References: <6DF262F0BAB5944AB03D1FE91354A1EF05FFE148AE@HICGWSEX01.ad.harman.com> Message-ID: <1253099150.16797.151.camel@localhost> The proper pipeline is ... gst-launch-0.10 playbin2 uri=rtsp://rtsp2.youtube.com/... Don't try building the pipeline yourself, playbin2 can do it for you On Tue, 2009-09-15 at 18:44 -0500, Gao, Ping wrote: > Hi all: > > > > Has anybody developed pipeline to play YouTube video (MPEG4 video and > AAC audio) through gstreamer rtspsrc, etc.? > > > > I am running on FC9 with the following packages installed: > > gstreamer-0.10.24 > > gst-plugins-base-0.10.24 > > gst-plugins-good-0.10.8 > > gst-plugins-bad-0.10.14 > > gst-plugins-ugly-0.10.12 > > gst-ffmpeg-0.10.8 > > > > > > Here is the command line I used and the error messages: > > [root at localhost zgrviewer]# gst-launch rtspsrc > location="rtsp://rtsp2.youtube.com/CiQLENy73wIaGwnYRKJ3bPTBdBMYESARFEgGUghzdGFuZGFyZAw=/0/0/0/video.3gp" ! rtph264depay ! ffdec_h264 ! xvimagesinkSetting pipeline to PAUSED ... > > Pipeline is live and does not need PREROLL ... > > Setting pipeline to PLAYING ... > > New clock: GstSystemClock > > ERROR: from > element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc2: > Internal data flow error. > > Additional debug info: > > gstbasesrc.c(2378): gst_base_src_loop > (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc2: > > streaming task paused, reason not-linked (-1) > > Execution ended after 2758843655 ns. > > Setting pipeline to PAUSED ... > > Setting pipeline to READY ... > > Setting pipeline to NULL ... > > Freeing pipeline ... > > > > > > What is missing in my command? If anybody has some sample application > code for YOuTuve video playback, that will help too. Thanks > > > > Ping > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From ping.gao at harman.com Wed Sep 16 18:40:54 2009 From: ping.gao at harman.com (Gao, Ping) Date: Wed, 16 Sep 2009 11:40:54 -0500 Subject: [gst-devel] What is the proper pipeline for YouTube video playback? In-Reply-To: <1253099150.16797.151.camel@localhost> Message-ID: <6DF262F0BAB5944AB03D1FE91354A1EF05FFE1510E@HICGWSEX01.ad.harman.com> Hi Edward: Thanks for your suggestion. I am preparing youtube playback on a device eventually. Since on the device playbin is not supported, I need to figure out the actual pipeline and write in software. I did try playbin myself before and it does give me video, however, I can't use that approach. Do you know a way to get the pipeline for this case? Thanks a lot. Ping -----Original Message----- From: Edward Hervey [mailto:bilboed at gmail.com] Sent: Wednesday, September 16, 2009 4:06 AM To: Discussion of the development of GStreamer Subject: Re: [gst-devel] What is the proper pipeline for YouTube video playback? The proper pipeline is ... gst-launch-0.10 playbin2 uri=rtsp://rtsp2.youtube.com/... Don't try building the pipeline yourself, playbin2 can do it for you On Tue, 2009-09-15 at 18:44 -0500, Gao, Ping wrote: > Hi all: > > > > Has anybody developed pipeline to play YouTube video (MPEG4 video and > AAC audio) through gstreamer rtspsrc, etc.? > > > > I am running on FC9 with the following packages installed: > > gstreamer-0.10.24 > > gst-plugins-base-0.10.24 > > gst-plugins-good-0.10.8 > > gst-plugins-bad-0.10.14 > > gst-plugins-ugly-0.10.12 > > gst-ffmpeg-0.10.8 > > > > > > Here is the command line I used and the error messages: > > [root at localhost zgrviewer]# gst-launch rtspsrc > location="rtsp://rtsp2.youtube.com/CiQLENy73wIaGwnYRKJ3bPTBdBMYESARFEgGUghzdGFuZGFyZAw=/0/0/0/video.3gp" ! rtph264depay ! ffdec_h264 ! xvimagesinkSetting pipeline to PAUSED ... > > Pipeline is live and does not need PREROLL ... > > Setting pipeline to PLAYING ... > > New clock: GstSystemClock > > ERROR: from > element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc2: > Internal data flow error. > > Additional debug info: > > gstbasesrc.c(2378): gst_base_src_loop > (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc2: > > streaming task paused, reason not-linked (-1) > > Execution ended after 2758843655 ns. > > Setting pipeline to PAUSED ... > > Setting pipeline to READY ... > > Setting pipeline to NULL ... > > Freeing pipeline ... > > > > > > What is missing in my command? If anybody has some sample application > code for YOuTuve video playback, that will help too. Thanks > > > > Ping > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel ------------------------------------------------------------------------------ Come build with us! The BlackBerry® Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9-12, 2009. Register now! http://p.sf.net/sfu/devconf _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From shawn_mcmurdo at yahoo.com Wed Sep 16 18:54:54 2009 From: shawn_mcmurdo at yahoo.com (Shawn McMurdo) Date: Wed, 16 Sep 2009 09:54:54 -0700 (PDT) Subject: [gst-devel] What is the proper pipeline for YouTube video playback? In-Reply-To: <6DF262F0BAB5944AB03D1FE91354A1EF05FFE1510E@HICGWSEX01.ad.harman.com> References: <6DF262F0BAB5944AB03D1FE91354A1EF05FFE1510E@HICGWSEX01.ad.harman.com> Message-ID: <666777.53562.qm@web53610.mail.re2.yahoo.com> If playbin works, you could write a short program that used playbin and then inspect the pipeline that gets created inside playbin. Shawn ________________________________ From: "Gao, Ping" To: Discussion of the development of GStreamer Sent: Wednesday, September 16, 2009 9:40:54 AM Subject: Re: [gst-devel] What is the proper pipeline for YouTube video playback? Hi Edward: Thanks for your suggestion. I am preparing youtube playback on a device eventually. Since on the device playbin is not supported, I need to figure out the actual pipeline and write in software. I did try playbin myself before and it does give me video, however, I can't use that approach. Do you know a way to get the pipeline for this case? Thanks a lot. Ping -----Original Message----- From: Edward Hervey [mailto:bilboed at gmail.com] Sent: Wednesday, September 16, 2009 4:06 AM To: Discussion of the development of GStreamer Subject: Re: [gst-devel] What is the proper pipeline for YouTube video playback? The proper pipeline is ... gst-launch-0.10 playbin2 uri=rtsp://rtsp2.youtube.com/... Don't try building the pipeline yourself, playbin2 can do it for you On Tue, 2009-09-15 at 18:44 -0500, Gao, Ping wrote: > Hi all: > > > > Has anybody developed pipeline to play YouTube video (MPEG4 video and > AAC audio) through gstreamer rtspsrc, etc.? > > > > I am running on FC9 with the following packages installed: > > gstreamer-0.10.24 > > gst-plugins-base-0.10.24 > > gst-plugins-good-0.10.8 > > gst-plugins-bad-0.10.14 > > gst-plugins-ugly-0.10.12 > > gst-ffmpeg-0.10.8 > > > > > > Here is the command line I used and the error messages: > > [root at localhost zgrviewer]# gst-launch rtspsrc > location="rtsp://rtsp2.youtube.com/CiQLENy73wIaGwnYRKJ3bPTBdBMYESARFEgGUghzdGFuZGFyZAw=/0/0/0/video.3gp" ! rtph264depay ! ffdec_h264 ! xvimagesinkSetting pipeline to PAUSED ... > > Pipeline is live and does not need PREROLL ... > > Setting pipeline to PLAYING ... > > New clock: GstSystemClock > > ERROR: from > element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc2: > Internal data flow error. > > Additional debug info: > > gstbasesrc.c(2378): gst_base_src_loop > (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc2: > > streaming task paused, reason not-linked (-1) > > Execution ended after 2758843655 ns. > > Setting pipeline to PAUSED ... > > Setting pipeline to READY ... > > Setting pipeline to NULL ... > > Freeing pipeline ... > > > > > > What is missing in my command? If anybody has some sample application > code for YOuTuve video playback, that will help too. Thanks > > > > Ping > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel ------------------------------------------------------------------------------ Come build with us! The BlackBerry® Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9-12, 2009. Register now! http://p.sf.net/sfu/devconf _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel ------------------------------------------------------------------------------ Come build with us! The BlackBerry® Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9-12, 2009. Register now! http://p.sf.net/sfu/devconf _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: From walter.kulecz-1 at nasa.gov Wed Sep 16 20:03:46 2009 From: walter.kulecz-1 at nasa.gov (Kulecz, Walter (JSC-SK)[WYLE LABORATORIES]) Date: Wed, 16 Sep 2009 13:03:46 -0500 Subject: [gst-devel] What is the proper pipeline for YouTube video playback? In-Reply-To: <6DF262F0BAB5944AB03D1FE91354A1EF05FFE1510E@HICGWSEX01.ad.harman.com> References: <1253099150.16797.151.camel@localhost>, <6DF262F0BAB5944AB03D1FE91354A1EF05FFE1510E@HICGWSEX01.ad.harman.com> Message-ID: <1F2262C21D3BB74DBE5C9576EAB4BF4F944D3AC7E2@NDJSSCC03.ndc.nasa.gov> Maybe a dumb question, but if playbin is not supported on your device, perhaps its because some of the other bits and pieces it needs aren't there either? --wally. ________________________________________ From: Gao, Ping [ping.gao at harman.com] Sent: Wednesday, September 16, 2009 11:40 AM To: Discussion of the development of GStreamer Subject: Re: [gst-devel] What is the proper pipeline for YouTube video playback? Hi Edward: Thanks for your suggestion. I am preparing youtube playback on a device eventually. Since on the device playbin is not supported, I need to figure out the actual pipeline and write in software. I did try playbin myself before and it does give me video, however, I can't use that approach. Do you know a way to get the pipeline for this case? Thanks a lot. Ping From ping.gao at harman.com Wed Sep 16 20:25:39 2009 From: ping.gao at harman.com (Gao, Ping) Date: Wed, 16 Sep 2009 13:25:39 -0500 Subject: [gst-devel] What is the proper pipeline for YouTube video playback? In-Reply-To: <1F2262C21D3BB74DBE5C9576EAB4BF4F944D3AC7E2@NDJSSCC03.ndc.nasa.gov> Message-ID: <6DF262F0BAB5944AB03D1FE91354A1EF05FFE50635@HICGWSEX01.ad.harman.com> It does support gstreamer-0.10.22, gst-plugins-base-0.0.10.22, and part of gst-plugins-good-0.10.14. I already made rtsp build working for my device. My plan was to have youtube video play on my Linux host first, then apply to my device. I know all the proper codecs are all available on my device, just different from those used in Linux host since they are supported in hardware. However, I'm having hard time to find the proper pipeline. -----Original Message----- From: Kulecz, Walter (JSC-SK)[WYLE LABORATORIES] [mailto:walter.kulecz-1 at nasa.gov] Sent: Wednesday, September 16, 2009 11:04 AM To: Discussion of the development of GStreamer Subject: Re: [gst-devel] What is the proper pipeline for YouTube video playback? Maybe a dumb question, but if playbin is not supported on your device, perhaps its because some of the other bits and pieces it needs aren't there either? --wally. ________________________________________ From: Gao, Ping [ping.gao at harman.com] Sent: Wednesday, September 16, 2009 11:40 AM To: Discussion of the development of GStreamer Subject: Re: [gst-devel] What is the proper pipeline for YouTube video playback? Hi Edward: Thanks for your suggestion. I am preparing youtube playback on a device eventually. Since on the device playbin is not supported, I need to figure out the actual pipeline and write in software. I did try playbin myself before and it does give me video, however, I can't use that approach. Do you know a way to get the pipeline for this case? Thanks a lot. Ping ------------------------------------------------------------------------------ Come build with us! The BlackBerry® Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9-12, 2009. Register now! http://p.sf.net/sfu/devconf _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From startoftext at gmail.com Wed Sep 16 20:54:58 2009 From: startoftext at gmail.com (James Pearson) Date: Wed, 16 Sep 2009 13:54:58 -0500 Subject: [gst-devel] alsasrc device=hw:1 pipeline problems In-Reply-To: <4AB0A594.80007@hora-obscura.de> References: <25458576.post@talk.nabble.com> <4AB0A594.80007@hora-obscura.de> Message-ID: <39C88F1C-9D32-487B-97D6-884C2C8AF7A4@gmail.com> Thanks for your suggestion. I did try the pipeline you suggested and it had no error. So then i tried the pipeline you suggested but replaced fakesink with filesink and it wrote an mp3 file that plays correctly. True i probably dont need that many queues but it still does not work without them. Any other suggestions? -James Pearson- startoftext at gmail.com 214-538-8929 On Sep 16, 2009, at 3:45 AM, Stefan Kost wrote: > startoftext schrieb: >> Can any one here tell me why this pipeline does not work? >> >> gst-launch-0.10 v4l2src ! queue ! ffmpegcolorspace ! ffenc_flv ! >> queue ! >> ffmux_flv name=mux alsasrc device=hw:1 ! queue ! audioconvert ! >> lame ! queue >> ! mux. mux. ! filesink location=vid.flv >> >> Here is the output i get from above: >> Setting pipeline to PAUSED ... >> Pipeline is live and does not need PREROLL ... >> Setting pipeline to PLAYING ... >> New clock: GstAudioSrcClock >> ERROR: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: >> Internal >> data flow error. >> Additional debug info: >> gstbasesrc.c(2334): gst_base_src_loop (): >> /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: >> streaming task paused, reason not-negotiated (-4) >> Execution ended after 37319285 ns. >> Setting pipeline to PAUSED ... >> Setting pipeline to READY ... >> Setting pipeline to NULL ... >> Freeing pipeline ... >> >> >> Funny thing is that if i change the device from hw:1 to hw:0 it >> works. Also >> this pipeline below DOES work. >> >> gst-launch-0.10 v4l2src ! queue ! xvimagesink . alsasrc device=hw:1 ! >> alsasink >> > what you need to get working is > > alsasrc device=hw:1 ! queue ! audioconvert ! lame ! fakesink > > I suspect sampling rates, so please try > > > alsasrc device=hw:1 ! queue ! audioresample ! audioconvert ! lame ! > fakesink > > Also consider to use less queues. Does that work? > > gst-launch-0.10 v4l2src ! ffmpegcolorspace ! ffenc_flv ! queue ! > ffmux_flv name=mux alsasrc device=hw:1 ! audioresample ! > audioconvert ! lame ! queue > ! mux. mux. ! filesink location=vid.flv > > Stefan > >> Any one have any ideas? >> > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart > your > developing skills, take BlackBerry mobile applications to market and > stay > ahead of the curve. Join us from November 9-12, 2009. Register > now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From gstelzz at yahoo.fr Wed Sep 16 21:15:14 2009 From: gstelzz at yahoo.fr (Aurelien Grimaud) Date: Wed, 16 Sep 2009 21:15:14 +0200 Subject: [gst-devel] What is the proper pipeline for YouTube video playback? In-Reply-To: <6DF262F0BAB5944AB03D1FE91354A1EF05FFE1510E@HICGWSEX01.ad.harman.com> References: <6DF262F0BAB5944AB03D1FE91354A1EF05FFE1510E@HICGWSEX01.ad.harman.com> Message-ID: <4AB13942.9030709@yahoo.fr> Hi, If you want to know what is used by playbin2, use the -v option ... gst-launch-0.10 -v playbin2 uri="rtsp://rtsp2.youtube.com/CiQLENy73wIaGwnYRKJ3bPTBdBMYESARFEgGUghzdGFuZGFyZAw=/0/0/0/video.3gp" Then you see this is mpeg4 ... So the correct pipeline is gst-launch-0.10 -v rtspsrc location="rtsp://rtsp2.youtube.com/CiQLENy73wIaGwnYRKJ3bPTBdBMYESARFEgGUghzdGFuZGFyZAw=/0/0/0/video.3gp" ! rtpmp4vdepay ! ffdec_mpeg4 ! xvimagesink Aurelien Le 16/09/2009 18:40, Gao, Ping a ?crit : > Hi Edward: > > Thanks for your suggestion. > I am preparing youtube playback on a device eventually. Since on the device playbin is not supported, I need to figure out the actual pipeline and write in software. I did try playbin myself before and it does give me video, however, I can't use that approach. Do you know a way to get the pipeline for this case? Thanks a lot. > > Ping > > -----Original Message----- > From: Edward Hervey [mailto:bilboed at gmail.com] > Sent: Wednesday, September 16, 2009 4:06 AM > To: Discussion of the development of GStreamer > Subject: Re: [gst-devel] What is the proper pipeline for YouTube video playback? > > The proper pipeline is ... > > gst-launch-0.10 playbin2 uri=rtsp://rtsp2.youtube.com/... > > Don't try building the pipeline yourself, playbin2 can do it for you > > On Tue, 2009-09-15 at 18:44 -0500, Gao, Ping wrote: > >> Hi all: >> >> >> >> Has anybody developed pipeline to play YouTube video (MPEG4 video and >> AAC audio) through gstreamer rtspsrc, etc.? >> >> >> >> I am running on FC9 with the following packages installed: >> >> gstreamer-0.10.24 >> >> gst-plugins-base-0.10.24 >> >> gst-plugins-good-0.10.8 >> >> gst-plugins-bad-0.10.14 >> >> gst-plugins-ugly-0.10.12 >> >> gst-ffmpeg-0.10.8 >> >> >> >> >> >> Here is the command line I used and the error messages: >> >> [root at localhost zgrviewer]# gst-launch rtspsrc >> location="rtsp://rtsp2.youtube.com/CiQLENy73wIaGwnYRKJ3bPTBdBMYESARFEgGUghzdGFuZGFyZAw=/0/0/0/video.3gp" ! rtph264depay ! ffdec_h264 ! xvimagesinkSetting pipeline to PAUSED ... >> >> Pipeline is live and does not need PREROLL ... >> >> Setting pipeline to PLAYING ... >> >> New clock: GstSystemClock >> >> ERROR: from >> element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc2: >> Internal data flow error. >> >> Additional debug info: >> >> gstbasesrc.c(2378): gst_base_src_loop >> (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc2: >> >> streaming task paused, reason not-linked (-1) >> >> Execution ended after 2758843655 ns. >> >> Setting pipeline to PAUSED ... >> >> Setting pipeline to READY ... >> >> Setting pipeline to NULL ... >> >> Freeing pipeline ... >> >> >> >> >> >> What is missing in my command? If anybody has some sample application >> code for YOuTuve video playback, that will help too. Thanks >> >> >> >> Ping >> >> >> ------------------------------------------------------------------------------ >> Come build with us! The BlackBerry® Developer Conference in SF, CA >> is the only developer event you need to attend this year. Jumpstart your >> developing skills, take BlackBerry mobile applications to market and stay >> ahead of the curve. Join us from November 9-12, 2009. Register now! >> http://p.sf.net/sfu/devconf >> _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From ensonic at hora-obscura.de Wed Sep 16 21:54:03 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Wed, 16 Sep 2009 22:54:03 +0300 Subject: [gst-devel] What is the proper pipeline for YouTube video playback? In-Reply-To: <6DF262F0BAB5944AB03D1FE91354A1EF05FFE1510E@HICGWSEX01.ad.harman.com> References: <6DF262F0BAB5944AB03D1FE91354A1EF05FFE1510E@HICGWSEX01.ad.harman.com> Message-ID: <4AB1425B.8020003@hora-obscura.de> Gao, Ping schrieb: > Hi Edward: > > Thanks for your suggestion. > I am preparing youtube playback on a device eventually. Since on the device playbin is not supported, I need to figure out the actual pipeline and write in software. I did try playbin myself before and it does give me video, however, I can't use that approach. Do you know a way to get the pipeline for this case? Thanks a lot. Can you tell use why you cannot use playbin or even better playbin2. It is supposed to work fine on embedded devices too. And if it does not for you then it might be worth to figure that out and fix it. Stefan > > Ping > > -----Original Message----- > From: Edward Hervey [mailto:bilboed at gmail.com] > Sent: Wednesday, September 16, 2009 4:06 AM > To: Discussion of the development of GStreamer > Subject: Re: [gst-devel] What is the proper pipeline for YouTube video playback? > > The proper pipeline is ... > > gst-launch-0.10 playbin2 uri=rtsp://rtsp2.youtube.com/... > > Don't try building the pipeline yourself, playbin2 can do it for you > > On Tue, 2009-09-15 at 18:44 -0500, Gao, Ping wrote: >> Hi all: >> >> >> >> Has anybody developed pipeline to play YouTube video (MPEG4 video and >> AAC audio) through gstreamer rtspsrc, etc.? >> >> >> >> I am running on FC9 with the following packages installed: >> >> gstreamer-0.10.24 >> >> gst-plugins-base-0.10.24 >> >> gst-plugins-good-0.10.8 >> >> gst-plugins-bad-0.10.14 >> >> gst-plugins-ugly-0.10.12 >> >> gst-ffmpeg-0.10.8 >> >> >> >> >> >> Here is the command line I used and the error messages: >> >> [root at localhost zgrviewer]# gst-launch rtspsrc >> location="rtsp://rtsp2.youtube.com/CiQLENy73wIaGwnYRKJ3bPTBdBMYESARFEgGUghzdGFuZGFyZAw=/0/0/0/video.3gp" ! rtph264depay ! ffdec_h264 ! xvimagesinkSetting pipeline to PAUSED ... >> >> Pipeline is live and does not need PREROLL ... >> >> Setting pipeline to PLAYING ... >> >> New clock: GstSystemClock >> >> ERROR: from >> element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc2: >> Internal data flow error. >> >> Additional debug info: >> >> gstbasesrc.c(2378): gst_base_src_loop >> (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc2: >> >> streaming task paused, reason not-linked (-1) >> >> Execution ended after 2758843655 ns. >> >> Setting pipeline to PAUSED ... >> >> Setting pipeline to READY ... >> >> Setting pipeline to NULL ... >> >> Freeing pipeline ... >> >> >> >> >> >> What is missing in my command? If anybody has some sample application >> code for YOuTuve video playback, that will help too. Thanks >> >> >> >> Ping >> >> >> ------------------------------------------------------------------------------ >> Come build with us! The BlackBerry® Developer Conference in SF, CA >> is the only developer event you need to attend this year. Jumpstart your >> developing skills, take BlackBerry mobile applications to market and stay >> ahead of the curve. Join us from November 9-12, 2009. Register now! >> http://p.sf.net/sfu/devconf >> _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From ping.gao at harman.com Wed Sep 16 21:59:23 2009 From: ping.gao at harman.com (Gao, Ping) Date: Wed, 16 Sep 2009 14:59:23 -0500 Subject: [gst-devel] What is the proper pipeline for YouTube video playback? In-Reply-To: <4AB13942.9030709@yahoo.fr> Message-ID: <6DF262F0BAB5944AB03D1FE91354A1EF05FFE5091A@HICGWSEX01.ad.harman.com> Great. How do I add the audio then? -----Original Message----- From: Aurelien Grimaud [mailto:gstelzz at yahoo.fr] Sent: Wednesday, September 16, 2009 12:15 PM To: Discussion of the development of GStreamer Subject: Re: [gst-devel] What is the proper pipeline for YouTube video playback? Hi, If you want to know what is used by playbin2, use the -v option ... gst-launch-0.10 -v playbin2 uri="rtsp://rtsp2.youtube.com/CiQLENy73wIaGwnYRKJ3bPTBdBMYESARFEgGUghzdGFuZGFyZAw=/0/0/0/video.3gp" Then you see this is mpeg4 ... So the correct pipeline is gst-launch-0.10 -v rtspsrc location="rtsp://rtsp2.youtube.com/CiQLENy73wIaGwnYRKJ3bPTBdBMYESARFEgGUghzdGFuZGFyZAw=/0/0/0/video.3gp" ! rtpmp4vdepay ! ffdec_mpeg4 ! xvimagesink Aurelien Le 16/09/2009 18:40, Gao, Ping a ?crit : > Hi Edward: > > Thanks for your suggestion. > I am preparing youtube playback on a device eventually. Since on the device playbin is not supported, I need to figure out the actual pipeline and write in software. I did try playbin myself before and it does give me video, however, I can't use that approach. Do you know a way to get the pipeline for this case? Thanks a lot. > > Ping > > -----Original Message----- > From: Edward Hervey [mailto:bilboed at gmail.com] > Sent: Wednesday, September 16, 2009 4:06 AM > To: Discussion of the development of GStreamer > Subject: Re: [gst-devel] What is the proper pipeline for YouTube video playback? > > The proper pipeline is ... > > gst-launch-0.10 playbin2 uri=rtsp://rtsp2.youtube.com/... > > Don't try building the pipeline yourself, playbin2 can do it for you > > On Tue, 2009-09-15 at 18:44 -0500, Gao, Ping wrote: > >> Hi all: >> >> >> >> Has anybody developed pipeline to play YouTube video (MPEG4 video and >> AAC audio) through gstreamer rtspsrc, etc.? >> >> >> >> I am running on FC9 with the following packages installed: >> >> gstreamer-0.10.24 >> >> gst-plugins-base-0.10.24 >> >> gst-plugins-good-0.10.8 >> >> gst-plugins-bad-0.10.14 >> >> gst-plugins-ugly-0.10.12 >> >> gst-ffmpeg-0.10.8 >> >> >> >> >> >> Here is the command line I used and the error messages: >> >> [root at localhost zgrviewer]# gst-launch rtspsrc >> location="rtsp://rtsp2.youtube.com/CiQLENy73wIaGwnYRKJ3bPTBdBMYESARFEgGUghzdGFuZGFyZAw=/0/0/0/video.3gp" ! rtph264depay ! ffdec_h264 ! xvimagesinkSetting pipeline to PAUSED ... >> >> Pipeline is live and does not need PREROLL ... >> >> Setting pipeline to PLAYING ... >> >> New clock: GstSystemClock >> >> ERROR: from >> element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc2: >> Internal data flow error. >> >> Additional debug info: >> >> gstbasesrc.c(2378): gst_base_src_loop >> (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc2: >> >> streaming task paused, reason not-linked (-1) >> >> Execution ended after 2758843655 ns. >> >> Setting pipeline to PAUSED ... >> >> Setting pipeline to READY ... >> >> Setting pipeline to NULL ... >> >> Freeing pipeline ... >> >> >> >> >> >> What is missing in my command? If anybody has some sample application >> code for YOuTuve video playback, that will help too. Thanks >> >> >> >> Ping >> >> >> ------------------------------------------------------------------------------ >> Come build with us! The BlackBerry® Developer Conference in SF, CA >> is the only developer event you need to attend this year. Jumpstart your >> developing skills, take BlackBerry mobile applications to market and stay >> ahead of the curve. Join us from November 9-12, 2009. Register now! >> http://p.sf.net/sfu/devconf >> _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ------------------------------------------------------------------------------ Come build with us! The BlackBerry® Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9-12, 2009. Register now! http://p.sf.net/sfu/devconf _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From yiliangb at gmail.com Thu Sep 17 01:29:47 2009 From: yiliangb at gmail.com (Yiliang Bao) Date: Wed, 16 Sep 2009 16:29:47 -0700 (PDT) Subject: [gst-devel] Use RTSP server to transmit the video stream in one branch of pipeline. Message-ID: <25482745.post@talk.nabble.com> Hi, I have the following pipeline. The video is being locally captured continuously, and at the same time, I need to transmit the video through RTP based on user request by dynamically switching on and off the second branch of the tee. gst-launch v4l2src ! x264enc ! tee name=t ! queue ! filesink location=capture.264 t. ! queue ! mpegtsmux ! rtpmp2tpay ! udpsink host=192.168.1.64 port=5000 My question is: "Is it possible that I can transmit the second branch using gstreamer RTSP server?" Or "Is there any way that I can implement that functionality with gstreamer RTSP server/" I read the examples of RTSP server, it seems that I always need to register the complete pipeline with RTSP server, before the server starts. And the pipeline is started inside RTSP server. In this case, the local capture will start and stop with streaming. This is not desired. Thanks, Yiliang -- View this message in context: http://www.nabble.com/Use-RTSP-server-to-transmit-the-video-stream-in-one-branch-of-pipeline.-tp25482745p25482745.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From yiliangb at gmail.com Thu Sep 17 01:30:54 2009 From: yiliangb at gmail.com (Yiliang Bao) Date: Wed, 16 Sep 2009 16:30:54 -0700 (PDT) Subject: [gst-devel] How to use RTSP server to transmit live video stream in one branch of pipeline? Message-ID: <25482745.post@talk.nabble.com> Hi, I have the following pipeline. The video is being locally captured continuously, and at the same time, I need to transmit the video through RTP based on user request by dynamically switching on and off the second branch of the tee. gst-launch v4l2src ! x264enc ! tee name=t ! queue ! filesink location=capture.264 t. ! queue ! mpegtsmux ! rtpmp2tpay ! udpsink host=192.168.1.64 port=5000 My question is: "Is it possible that I can transmit the second branch using gstreamer RTSP server?" Or "Is there any way that I can implement that functionality with gstreamer RTSP server/" I read the examples of RTSP server, it seems that I always need to register the complete pipeline with RTSP server, before the server starts. And the pipeline is started inside RTSP server. In this case, the local capture will start and stop with streaming. This is not desired. Thanks, Yiliang -- View this message in context: http://www.nabble.com/How-to-use-RTSP-server-to-transmit-live-video-stream-in-one-branch-of-pipeline--tp25482745p25482745.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From jaisena at yahoo.com Thu Sep 17 01:37:23 2009 From: jaisena at yahoo.com (jayasena s) Date: Wed, 16 Sep 2009 16:37:23 -0700 (PDT) Subject: [gst-devel] how to generate a custom event In-Reply-To: <562003.1939.qm@web55303.mail.re4.yahoo.com> Message-ID: <601189.92053.qm@web55305.mail.re4.yahoo.com> Hi Jan, ?How I can send a custom event at xvimagesink and send it an element in the pipeline. Thanks, Jayasena --- On Sat, 9/12/09, jayasena s wrote: From: jayasena s Subject: Re: [gst-devel] A/V sync and stutter issue To: "Discussion of the development of GStreamer" Date: Saturday, September 12, 2009, 8:05 PM Hi Jan, Thanks for the suggestions. ? I would?start debugging the pipeline with gst-launch and totem. ? Also, Would like to know,?? When a buffer with decoded frame is sent using gst_pad_push() to downstream to?the xvimagesink , is there any event notification sent back with gst_pad_push() ?to the upstream elements, when the frame gets rendered at xvimagesink. If not with gst_pad_push(), is there any other API which can be used to send notification to the upstream element, after frame is rendered to the display from the xvimagesink. ? Thanks, Jayasena ? --- On Fri, 9/11/09, Jan Schmidt wrote: From: Jan Schmidt Subject: Re: [gst-devel] A/V sync and stutter issue To: "Discussion of the development of GStreamer" Date: Friday, September 11, 2009, 11:02 PM Hi, On Fri, 2009-09-11 at 11:22 +0100, Jan Schmidt wrote: Hrmn, my initial reply was lacking somewhat in detail ;) > On Thu, 2009-09-10 at 11:43 -0700, jayasena s wrote: > >? *snip* > >? ? ? ???Hi, > >? ? ? ? ? I am having A/V Sync and stutter issues , when using totem > >? ? ? ???player with gstreamer. > >? ? ? ???With gst-launch, audio and video are in sync and video is > >? ? ? ???smooth ( no stutter issue). > >? ? ? ? ? > >? ? ? ???Timestamps seem to be good, the stutter is more visible for > >? ? ? ???the first 4 seconds of the playback. > >? ? ? ? ? > >? ? ? ???Did anyone come across this issue, Could you suggest ways to > >? ? ? ???debug these issues I'd start by identifying the difference between the pipeline you're giving gst-launch and what totem is using. Totem (depending on the version) is using either playbin or playbin2, with the gconf audio and video sinks. Cheers, Jan. > >? ? ? ? ? > >? ? ? ???Thanks, > >? ? ? ???Jai > >? ? ? ??? > >? ? ? ??? > >? ? ? ??? > >? ? ? ???-----Inline Attachment Follows----- > >? ? ? ??? > >? ? ? ???------------------------------------------------------------------------------ > >? ? ? ???Let Crystal Reports handle the reporting - Free Crystal > >? ? ? ???Reports 2008 30-Day > >? ? ? ???trial. Simplify your report design, integration and deployment > >? ? ? ???- and focus on > >? ? ? ???what you do best, core application coding. Discover what's new > >? ? ? ???with > >? ? ? ???Crystal Reports now.? http://p.sf.net/sfu/bobj-july > >? ? ? ??? > >? ? ? ???-----Inline Attachment Follows----- > >? ? ? ??? > >? ? ? ???_______________________________________________ > >? ? ? ???gstreamer-devel mailing list > >? ? ? ???gstreamer-devel at lists.sourceforge.net > >? ? ? ???https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > >? ? ? ??? > > > > ------------------------------------------------------------------------------ > > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day > > trial. Simplify your report design, integration and deployment - and focus on > > what you do best, core application coding. Discover what's new with > > Crystal Reports now.? http://p.sf.net/sfu/bobj-july > > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -- Jan Schmidt ------------------------------------------------------------------------------ Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day trial. Simplify your report design, integration and deployment - and focus on what you do best, core application coding. Discover what's new with Crystal Reports now.? http://p.sf.net/sfu/bobj-july _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -----Inline Attachment Follows----- ------------------------------------------------------------------------------ Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day trial. Simplify your report design, integration and deployment - and focus on what you do best, core application coding. Discover what's new with Crystal Reports now.? http://p.sf.net/sfu/bobj-july -----Inline Attachment Follows----- _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: From shawn_mcmurdo at yahoo.com Thu Sep 17 01:47:14 2009 From: shawn_mcmurdo at yahoo.com (Shawn McMurdo) Date: Wed, 16 Sep 2009 16:47:14 -0700 (PDT) Subject: [gst-devel] Why audio extract on flv won't preroll? In-Reply-To: <818311.52191.qm@web53608.mail.re2.yahoo.com> References: <818311.52191.qm@web53608.mail.re2.yahoo.com> Message-ID: <203056.57086.qm@web53610.mail.re2.yahoo.com> Nobody knows? Can anyone at least comment on whether my pipeline is correct? Thanks! Shawn ________________________________ From: Shawn McMurdo To: Gstreamer Devel Sent: Monday, September 14, 2009 5:19:13 PM Subject: [gst-devel] Why audio extract on flv won't preroll? Hi, I am trying to do an audio extract on various types of videos. Using decodebin as shown below on this video: http://www.shawnim.com/ax/balboa.flv causes gstreamer to hang in the preroll. The same pipeline works on other flvs and wmvs. Can anyone explain why this is happening? I am running gstreamer 0.10.24 on Fedora 11 (2.6.29.6-217.2.16.fc11.x86_64). The pipeline I am using is: gst-launch-0.10 -m -t -v filesrc location=$1 ! decodebin ! audioconvert ! audio/x-raw-int,channels=1,width=16,depth=16 ! audioresample ! audio/x-raw-int,rate=16000 ! wavenc ! filesink location=$1.wav The output from running this pipeline is: http://www.shawnim.com/ax/balboa.out BTW, decodebin2 works on this file but seems to have trouble with other files. Any explanations or pointer appreciated. Thanks. Shawn -------------- next part -------------- An HTML attachment was scrubbed... URL: From thaytan at noraisin.net Thu Sep 17 02:49:30 2009 From: thaytan at noraisin.net (Jan Schmidt) Date: Thu, 17 Sep 2009 01:49:30 +0100 Subject: [gst-devel] Core/Base/Python/FFmpeg - 1st pre-releases In-Reply-To: <1252714620.13334.38.camel@fancy> References: <1252714620.13334.38.camel@fancy> Message-ID: <1253148570.5055.9.camel@fancy> Hi guys, Some new pre-release tarballs are available for core, base and gst-python. There's no new tarball for gst-ffmpeg at the moment. The current pre-release set is: http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.24.3.tar.bz2 http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.24.3.tar.bz2 http://gstreamer.freedesktop.org/src/gst-python/pre/gst-python-0.10.16.3.tar.bz2 http://gstreamer.freedesktop.org/src/gst-ffmpeg/pre/gst-ffmpeg-0.10.8.2.tar.bz2 md5sums: 1c263bfa047929143b70256268b7c2ba gstreamer-0.10.24.3.tar.bz2 2bbea786578ef38e7e5bb1ba08ea18f7 gst-plugins-base-0.10.24.3.tar.bz2 2bbea786578ef38e7e5bb1ba08ea18f7 gst-python-0.10.16.3.tar.bz2 e6057902f39301eb2d08d48b34a91dd6 gst-ffmpeg-0.10.8.2.tar.bz2 Changes in core are: * utils: Fix GMP scaling unit test (bug #595133) * Fix out-of-tree build * docs: GST_MESSAGE_STREAM_STATUS is implemented nowadays. * introspection: Build pkgconfig before all libraries and set PKG_CONFIG_PATH * introspection: Don't typedef GstTagList to GstStructure for gobject-introspe Changes in base: * vorbistag: don't ever return NULL in list of strings. * playsink: Expose mute,volume,vis-plugin and font-desc properties * GstPlaySink: Expose 'reconfigure' as an action signal. * GstPlaySink: Expose flags as a gobject property. * playback: Register playsink as an element. * docs: add new gst_stream_volume_get_type to types file * oggdemux: Fix duration calculation for truncated files * introspection: Build pkgconfig before all libraries and set PKG_CONFIG_PATH * theoraenc: Fix a string leak in _getcaps() Changes in the python bindings: * Update definitions and ignores for core/base 0.10.25. Fixes #587432 Please test with your favourite programs, and file bugs for any problems in http://bugzilla.gnome.org/ Cheers, Jan. On Sat, 2009-09-12 at 01:17 +0100, Jan Schmidt wrote: > Hi all, > > All four of the modules in this release cycle are now frozen for > commits, and the 1st pre-releases are available. > > The tarballs are available at: > http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.24.2.tar.bz2 > http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.24.2.tar.bz2 > http://gstreamer.freedesktop.org/src/gst-python/pre/gst-python-0.10.16.2.tar.bz2 > http://gstreamer.freedesktop.org/src/gst-ffmpeg/pre/gst-ffmpeg-0.10.8.2.tar.bz2 > > md5sums: > 851fe968803b8a49fdac513435142f0b gstreamer-0.10.24.2.tar.bz2 > 05cd5878ad4182c803c821cd053fdcce gst-plugins-base-0.10.24.2.tar.bz2 > e513c18c22665146eca98a2dd390b42a gst-python-0.10.16.2.tar.bz2 > e6057902f39301eb2d08d48b34a91dd6 gst-ffmpeg-0.10.8.2.tar.bz2 > > Please give them a good test and report any problems, and especially > regressions, in bugzilla - http://bugzilla.gnome.org/ > > See http://gstreamer.freedesktop.org/wiki/ReleasePlanning2009-2 for more > details about the release schedule. > > Cheers, > Jan. -- Jan Schmidt From kpawan at gmail.com Thu Sep 17 07:53:36 2009 From: kpawan at gmail.com (Kumar, Pawan) Date: Thu, 17 Sep 2009 11:23:36 +0530 Subject: [gst-devel] Gstreamer pipeline to convert from avi to mp4 format !! Message-ID: Hi All, I am looking for a Gstreamer pipeline to convert from avi to mp4 format. Here mp4 should contain the same the elementary Audio and Video streams as stored in avi. test_h264_aac.avi : (Video.h264, Audio.aac) -----------> output_h264_aac.mp4 : (Video.h264, Audio.aac) So I do not want to decode the elementary streams and again encode it back before mux-ing with the MP4 I tried with the below pipeline. gst-launch-0.10 filesrc location=test_h264_aac.avi ! avidemux name=demux { qtmux name=mux ! filesink location=output.mp4 } { demux. ! queue ! audiopass ! mux. } { demux. ! queue ! videopass ! mux. } Here audiopass, and videopass are gstreamer plugins, which pass through the audio and video elementary streams as it. These elements are used only for caps negotiation of the audio and video src pads of AVI demuxer with the audio and video sink pads of the MP4 muxer. This pipeline does not write out any Data. Can somebody please suggest me a pipeline to do the conversion mentioned above ? Thanks, /Pawan -------------- next part -------------- An HTML attachment was scrubbed... URL: From gstelzz at yahoo.fr Thu Sep 17 09:27:59 2009 From: gstelzz at yahoo.fr (Aurelien Grimaud) Date: Thu, 17 Sep 2009 09:27:59 +0200 Subject: [gst-devel] Gstreamer pipeline to convert from avi to mp4 format !! In-Reply-To: References: Message-ID: <4AB1E4FF.3070909@yahoo.fr> Hi, Could you post the output of your gst-launch ? What does "does not write out any data" means ? Stuck in preroll ?, not negotiated ? Did you try to GST_DEBUG it ? Should give hints on what happens ... Aurelien Kumar, Pawan a ?crit : > > Hi All, > > I am looking for a Gstreamer pipeline to convert from avi to mp4 > format. Here mp4 should contain the same the elementary Audio and > Video streams as stored in avi. > > test_h264_aac.avi : (Video.h264, Audio.aac) -----------> > output_h264_aac.mp4 : (Video.h264, Audio.aac) > > So I do not want to decode the elementary streams and again encode > it back before mux-ing with the MP4 > > I tried with the below pipeline. > > gst-launch-0.10 filesrc location=test_h264_aac.avi ! avidemux > name=demux { qtmux name=mux ! filesink location=output.mp4 } { demux. > ! queue ! audiopass ! mux. } { demux. ! queue ! videopass ! mux. } > > Here audiopass, and videopass are gstreamer plugins, which pass > through the audio and video elementary streams as it. These elements > are used only for caps negotiation of the audio and video src pads of > AVI demuxer with the audio and video sink pads of the MP4 muxer. > > This pipeline does not write out any Data. > > Can somebody please suggest me a pipeline to do the conversion > mentioned above ? > > > Thanks, > /Pawan > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From dirk.griffioen at barcelonamedia.org Thu Sep 17 09:58:24 2009 From: dirk.griffioen at barcelonamedia.org (Dirk Griffioen) Date: Thu, 17 Sep 2009 09:58:24 +0200 Subject: [gst-devel] rtp pipeline in rtsp server, possible? Message-ID: <4AB1EC20.1040100@barcelonamedia.org> Hi, >From http://webcvs.freedesktop.org/gstreamer/gst-plugins-good/gst/rtp/README?view=markup I read the following: To correctly and completely use the RTP payloaders on the sender and the receiver you need to write an application. It is not possible to write a full blown RTP server with a single gst-launch line. That said, it is possible to do something functional with a few gst-launch lines. The biggest problem when constructing a correct gst-launch line lies on the receiver end. The receiver needs to know about the type of the RTP data along with a set of RTP configuration parameters. This information is usually transmitted to the client using some sort of session description language (SDP) over some reliable channel (HTTP/RTSP/...). As there is an RTSP server in Gstreamer, my question then is 'can I not run an rtp pipeline from inside this rtsp server'? That way I think it would work 'normally'. Secondly, how would I set this up? Any help much appreciated! Best, Dirk From dirk.griffioen at barcelonamedia.org Thu Sep 17 09:53:01 2009 From: dirk.griffioen at barcelonamedia.org (Dirk Griffioen) Date: Thu, 17 Sep 2009 09:53:01 +0200 Subject: [gst-devel] sdp broken? Message-ID: <4AB1EADD.9000109@barcelonamedia.org> Hi, A nice example on how to use rtp can be found here: http://ubuntuforums.org/showthread.php?t=882537 If I try the SDP variant, the stream stops after a few seconds; according to the forum because it is broken. I do have the impression that this is correct, so my question is 'is that correct'? (And how can this be fixed). Thanks in advance. Best, Dirk From dirk.griffioen at barcelonamedia.org Thu Sep 17 10:06:17 2009 From: dirk.griffioen at barcelonamedia.org (Dirk Griffioen) Date: Thu, 17 Sep 2009 10:06:17 +0200 Subject: [gst-devel] rtp streams stop unexpectedly Message-ID: <4AB1EDF9.5070104@barcelonamedia.org> Hi, I have the following 2 pipelines for streaming (send/receive) audio: *producer:* gst-launch -v gstrtpbin name=rtpbin \ filesrc location=filesrc location=~/Desktop/video.mp4 ! decodebin name=dec \ dec. ! queue ! audioresample ! audioconvert ! alawenc ! rtppcmapay ! rtpbin.send_rtp_sink_1 \ rtpbin.send_rtp_src_1 ! udpsink port=5002 host=127.0.0.1 ts-offset=0 \ rtpbin.send_rtcp_src_1 ! udpsink port=5003 host=127.0.0.1 sync=false async=false \ udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1 *consumer:* gst-launch -v gstrtpbin name=rtpbin latency=200 \ udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA" port=5002 ! rtpbin.recv_rtp_sink_1 \ rtpbin. ! rtppcmadepay ! decodebin ! audioconvert ! audioresample ! alsasink \ udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ rtpbin.send_rtcp_src_1 ! udpsink port=5007 host=127.0.0.1 sync=false async=false If I start the consumer and then the producer, the stream if fine. But if I then restart the producer, the consumer drops with the following message: /GstPipeline:pipeline0/GstRtpBin:rtpbin/GstRtpJitterBuffer:rtpjitterbuffer1.GstPad:src: caps = application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA /GstPipeline:pipeline0/GstRtpBin:rtpbin/GstRtpJitterBuffer:rtpjitterbuffer1.GstPad:sink: caps = application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA /GstPipeline:pipeline0/GstRtpBin:rtpbin/GstRtpPtDemux:rtpptdemux1.GstPad:sink: caps = application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA /GstPipeline:pipeline0/GstRtpBin:rtpbin.GstGhostPad:recv_rtp_src_1_1517622516_8: caps = application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA, payload=(int)8 /GstPipeline:pipeline0/GstRtpBin:rtpbin.GstGhostPad:recv_rtp_src_1_1517622516_8.GstProxyPad:proxypad6: caps = application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA, payload=(int)8 ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Internal data flow error. Additional debug info: gstbasesrc.c(2330): gst_base_src_loop (): /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: streaming task paused, reason not-linked (-1) Execution ended after 31686734760 ns. Is this expected behaviour? Can I somehow fix this, so I can start/stop the producer? Help is very much appreciated! Best, Dirk -------------- next part -------------- An HTML attachment was scrubbed... URL: From thaytan at noraisin.net Thu Sep 17 11:32:36 2009 From: thaytan at noraisin.net (Jan Schmidt) Date: Thu, 17 Sep 2009 10:32:36 +0100 Subject: [gst-devel] Core/Base/Python/FFmpeg - *2nd* pre-releases In-Reply-To: <1253148570.5055.9.camel@fancy> References: <1252714620.13334.38.camel@fancy> <1253148570.5055.9.camel@fancy> Message-ID: <1253179956.5055.12.camel@fancy> That was *2nd* pre-releases, not first :) J. On Thu, 2009-09-17 at 01:49 +0100, Jan Schmidt wrote: > Hi guys, > > Some new pre-release tarballs are available for core, base and > gst-python. There's no new tarball for gst-ffmpeg at the moment. > > The current pre-release set is: > > http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.24.3.tar.bz2 > http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.24.3.tar.bz2 > http://gstreamer.freedesktop.org/src/gst-python/pre/gst-python-0.10.16.3.tar.bz2 > http://gstreamer.freedesktop.org/src/gst-ffmpeg/pre/gst-ffmpeg-0.10.8.2.tar.bz2 > > md5sums: > 1c263bfa047929143b70256268b7c2ba gstreamer-0.10.24.3.tar.bz2 > 2bbea786578ef38e7e5bb1ba08ea18f7 gst-plugins-base-0.10.24.3.tar.bz2 > 2bbea786578ef38e7e5bb1ba08ea18f7 gst-python-0.10.16.3.tar.bz2 > e6057902f39301eb2d08d48b34a91dd6 gst-ffmpeg-0.10.8.2.tar.bz2 > > Changes in core are: > * utils: Fix GMP scaling unit test (bug #595133) > * Fix out-of-tree build > * docs: GST_MESSAGE_STREAM_STATUS is implemented nowadays. > * introspection: Build pkgconfig before all libraries and set PKG_CONFIG_PATH > * introspection: Don't typedef GstTagList to GstStructure for gobject-introspe > > Changes in base: > * vorbistag: don't ever return NULL in list of strings. > * playsink: Expose mute,volume,vis-plugin and font-desc properties > * GstPlaySink: Expose 'reconfigure' as an action signal. > * GstPlaySink: Expose flags as a gobject property. > * playback: Register playsink as an element. > * docs: add new gst_stream_volume_get_type to types file > * oggdemux: Fix duration calculation for truncated files > * introspection: Build pkgconfig before all libraries and set PKG_CONFIG_PATH > * theoraenc: Fix a string leak in _getcaps() > > Changes in the python bindings: > * Update definitions and ignores for core/base 0.10.25. Fixes #587432 > > Please test with your favourite programs, and file bugs for any problems in http://bugzilla.gnome.org/ > > Cheers, > Jan. > > On Sat, 2009-09-12 at 01:17 +0100, Jan Schmidt wrote: > > Hi all, > > > > All four of the modules in this release cycle are now frozen for > > commits, and the 1st pre-releases are available. > > > > The tarballs are available at: > > http://gstreamer.freedesktop.org/src/gstreamer/pre/gstreamer-0.10.24.2.tar.bz2 > > http://gstreamer.freedesktop.org/src/gst-plugins-base/pre/gst-plugins-base-0.10.24.2.tar.bz2 > > http://gstreamer.freedesktop.org/src/gst-python/pre/gst-python-0.10.16.2.tar.bz2 > > http://gstreamer.freedesktop.org/src/gst-ffmpeg/pre/gst-ffmpeg-0.10.8.2.tar.bz2 > > > > md5sums: > > 851fe968803b8a49fdac513435142f0b gstreamer-0.10.24.2.tar.bz2 > > 05cd5878ad4182c803c821cd053fdcce gst-plugins-base-0.10.24.2.tar.bz2 > > e513c18c22665146eca98a2dd390b42a gst-python-0.10.16.2.tar.bz2 > > e6057902f39301eb2d08d48b34a91dd6 gst-ffmpeg-0.10.8.2.tar.bz2 > > > > Please give them a good test and report any problems, and especially > > regressions, in bugzilla - http://bugzilla.gnome.org/ > > > > See http://gstreamer.freedesktop.org/wiki/ReleasePlanning2009-2 for more > > details about the release schedule. > > > > Cheers, > > Jan. -- Jan Schmidt From t.i.m at zen.co.uk Thu Sep 17 11:36:18 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Thu, 17 Sep 2009 10:36:18 +0100 Subject: [gst-devel] Why audio extract on flv won't preroll? In-Reply-To: <203056.57086.qm@web53610.mail.re2.yahoo.com> References: <818311.52191.qm@web53608.mail.re2.yahoo.com> <203056.57086.qm@web53610.mail.re2.yahoo.com> Message-ID: <1253180178.4950.6.camel@zingle> On Wed, 2009-09-16 at 16:47 -0700, Shawn McMurdo wrote: > Can anyone at least comment on whether my pipeline is correct? The pipeline is correct in principle. You might want the audioresample first and the audioconvert directly in front of wavenc though, because wavenc only accepts little endian pcm audio, so your pipeline might not work on big endian machines because audioresample only processes pcm audio in native endianness. That isn't related to your problem though. Decodebin2 works here because it uses multiqueue which handles buffering more correctly and dynamically for certain files, whereas decodebin uses more or less fixed size queues, which don't work for badly interleaved files and many HD video files. Recent versions of decodebin2 should work fine with all files, and I'm not aware of any problems with it. If you have files that don't work with it, please file a bug (or upgrade to a more recent gst-plugins-base if you're not using the latest), thanks! Cheers -Tim ________________________________________________________________________ > From: Shawn McMurdo > To: Gstreamer Devel > Sent: Monday, September 14, 2009 5:19:13 PM > Subject: [gst-devel] Why audio extract on flv won't preroll? > > Hi, > I am trying to do an audio extract on various types of videos. > > Using decodebin as shown below on this video: > http://www.shawnim.com/ax/balboa.flv > causes gstreamer to hang in the preroll. > > The same pipeline works on other flvs and wmvs. > Can anyone explain why this is happening? > > I am running gstreamer 0.10.24 on Fedora 11 > (2.6.29.6-217.2.16.fc11.x86_64). > The pipeline I am using is: > gst-launch-0.10 -m -t -v filesrc location=$1 ! decodebin ! > audioconvert ! audio/x-raw-int,channels=1,width=16,depth=16 ! > audioresample ! audio/x-raw-int,rate=16000 ! wavenc ! filesink > location=$1.wav > > The output from running this pipeline is: > http://www.shawnim.com/ax/balboa.out > > BTW, decodebin2 works on this file but seems to have trouble with > other files. > Any explanations or pointer appreciated. > Thanks. > Shawn > > > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From t.i.m at zen.co.uk Thu Sep 17 11:41:37 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Thu, 17 Sep 2009 10:41:37 +0100 Subject: [gst-devel] A/V sync and stutter issue In-Reply-To: <562003.1939.qm@web55303.mail.re4.yahoo.com> References: <562003.1939.qm@web55303.mail.re4.yahoo.com> Message-ID: <1253180497.4950.9.camel@zingle> On Sat, 2009-09-12 at 12:05 -0700, jayasena s wrote: > Also, Would like to know, When a buffer with decoded frame is sent > using gst_pad_push() to downstream to the xvimagesink , is there any > event notification sent back with gst_pad_push() to the upstream > elements, when the frame gets rendered at xvimagesink. If not with > gst_pad_push(), is there any other API which can be used to send > notification to the upstream element, after frame is rendered to the > display from the xvimagesink. A QoS event is usually sent upstream whenever a frame is rendered. Not sure how that helps you here though. Cheers -Tim From dxssx.dxssx at gmail.com Thu Sep 17 13:23:50 2009 From: dxssx.dxssx at gmail.com (dxssx) Date: Thu, 17 Sep 2009 19:23:50 +0800 Subject: [gst-devel] How can I slow down the sending speed of rtsp server Message-ID: Hi, I have encountered a problem when use rtsp-server as vod server. The rtsp server send stream too fast when I use a pipeline as below:./test-launch "filesrc location=a.m4v ! mpeg4videoparse ! rtpmp4vpay name=pay0" The pipeline runs fast on PC so that the server send stream much faster than the actual framerate of the file. But my player on another PC can not decode as fast as stream (almost 90Mbps). So I call for help here. Is there a way to slow down or control the speed of the server. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: From william at breezecapsule.com Thu Sep 17 16:02:31 2009 From: william at breezecapsule.com (William Lee) Date: Thu, 17 Sep 2009 15:02:31 +0100 Subject: [gst-devel] Clock Provider in a pipeline Message-ID: Hi all, I have a question regarding the clock selection in a pipeline. How does a pipeline select a clock if the pipeline contains multiple elements that are clock providers, for example multiple sinks? William -- William Lee @ Breeze Capsule Email: william at breezecapsule.com Mobile: +44 7984 368 182 -------------- next part -------------- An HTML attachment was scrubbed... URL: From jaisena at yahoo.com Thu Sep 17 17:52:02 2009 From: jaisena at yahoo.com (jayasena s) Date: Thu, 17 Sep 2009 08:52:02 -0700 (PDT) Subject: [gst-devel] A/V sync and stutter issue In-Reply-To: <1253180497.4950.9.camel@zingle> Message-ID: <36437.93543.qm@web55304.mail.re4.yahoo.com> Hi Tim, ?Thanks for the suggestion. I looked into the QOS events and? also found a way to send the event upstream using GST_EVENT_CUSTOM_UPSTREAM. THanks, Jayasena --- On Thu, 9/17/09, Tim-Philipp M?ller wrote: From: Tim-Philipp M?ller Subject: Re: [gst-devel] A/V sync and stutter issue To: gstreamer-devel at lists.sourceforge.net Date: Thursday, September 17, 2009, 10:41 AM On Sat, 2009-09-12 at 12:05 -0700, jayasena s wrote: > Also, Would like to know,???When a buffer with decoded frame is sent > using gst_pad_push() to downstream to the xvimagesink , is there any > event notification sent back with gst_pad_push()? to the upstream > elements, when the frame gets rendered at xvimagesink. If not with > gst_pad_push(), is there any other API which can be used to send > notification to the upstream element, after frame is rendered to the > display from the xvimagesink. A QoS event is usually sent upstream whenever a frame is rendered. Not sure how that helps you here though. Cheers -Tim ------------------------------------------------------------------------------ Come build with us! The BlackBerry® Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9-12, 2009. Register now! http://p.sf.net/sfu/devconf _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -------------- next part -------------- An HTML attachment was scrubbed... URL: From rob at ti.com Fri Sep 18 13:05:25 2009 From: rob at ti.com (Rob Clark) Date: Fri, 18 Sep 2009 06:05:25 -0500 Subject: [gst-devel] Using cairo/pixman for raw video in GStreamer In-Reply-To: References: Message-ID: On Sep 11, 2009, at 12:02 PM, Benjamin Otte wrote: > Hi, > > This is an idea that's been brewing in my head for a bit. After > thinking about it for a while and poking some people on IRC, I'm > pretty convinced it's the best way forward. so.. is the proposal that you would pass a cairo_surface_t in the GstBuffer (GstVideoBuffer?) Or would you continue to pass the buffer as just a byte ptr? I'm curious if the cairo dependency would only be for elements that are touching pixels? Or would platforms that have all the heavy lifting done on some sorts of hardware and/or coprocessor(s) also inherit this dependency? BR, -R From dirk.griffioen at barcelonamedia.org Fri Sep 18 13:26:01 2009 From: dirk.griffioen at barcelonamedia.org (Dirk Griffioen) Date: Fri, 18 Sep 2009 13:26:01 +0200 Subject: [gst-devel] multichannel vorbis Message-ID: <4AB36E49.2030102@barcelonamedia.org> Hi, Is there a way to create a multichannel vorbis file with gstreamer? I want to create a file (or a stream) with n discrete channels. I tried something like this: gst-launch -v vorbisenc name=v ! oggmux ! filesink location=sine.ogg \ audiotestsrc wave=sine ! audioconvert ! v. \ audiotestsrc wave=sine ! audioconvert ! v. But this does not work ... Thanks in advance. Best, Dirk From lrn1986 at gmail.com Fri Sep 18 14:57:17 2009 From: lrn1986 at gmail.com (LRN) Date: Fri, 18 Sep 2009 16:57:17 +0400 Subject: [gst-devel] multichannel vorbis In-Reply-To: <4AB36E49.2030102@barcelonamedia.org> References: <4AB36E49.2030102@barcelonamedia.org> Message-ID: <4AB383AD.2060704@gmail.com> Dirk Griffioen wrote: > Hi, > > Is there a way to create a multichannel vorbis file with gstreamer? I > want to create a file (or a stream) with n discrete channels. > > I tried something like this: > > gst-launch -v vorbisenc name=v ! oggmux ! filesink location=sine.ogg \ > audiotestsrc wave=sine ! audioconvert ! v. \ > audiotestsrc wave=sine ! audioconvert ! v. > > But this does not work ... > > Thanks in advance. > > Best, Dirk > I think you need to interleave several mono channels from audiotestsrc into a stereo stream and then shove them into the vorbis encoder. You need an audio mixer, or something like that. From costa_albert at yahoo.fr Fri Sep 18 16:18:40 2009 From: costa_albert at yahoo.fr (Albert Costa) Date: Fri, 18 Sep 2009 07:18:40 -0700 (PDT) Subject: [gst-devel] Real video framerate differs of given pipeline framerate Message-ID: <729870.19643.qm@web28408.mail.ukl.yahoo.com> Dear All, I have a question about usage of usb webcam. My pipeline is as follow: gst-launch-0.10.exe ksvideosrc ! ffmpegcolorspace ! videorate ! video/x-raw-yuv,width=640,framerate=30/1 ! ffmpegcolorspace ! clockoverlay ! ffmpegcolorspace ! ffenc_mpeg2video gop-size=1 quantizer=4 ! ffmux_mpegts ! filesink location=c:\\web.ts sync=false It connects to a webcam, overlays the clock, encode to mpeg ts format and save it to file. When I read the file, I can see that the frame timestamps are not in sync with reality for a bunch of seconds. I mean that the duration of a frame does not correspond to the "real" time elapsed (frame duration is set to 33ms, but in the real life maybe 43ms have been gone since last buffer was sent). This happens for about the 50 first frames. So playing the file afterwards makes the first frames be displayed faster than it should be (I use external player that bases himself on given framerate). No matter what I've tried (setting sync to 0/1, using or not videorate...) I always have this lag (because of camera init I guess). How can I ensure that the frames are generated at the real framerate? or how can I discard the frames that do not match the framerate? Best Regards, Al -------------- next part -------------- An HTML attachment was scrubbed... URL: From dirk.griffioen at barcelonamedia.org Fri Sep 18 16:33:52 2009 From: dirk.griffioen at barcelonamedia.org (Dirk Griffioen) Date: Fri, 18 Sep 2009 16:33:52 +0200 Subject: [gst-devel] multichannel vorbis In-Reply-To: <4AB383AD.2060704@gmail.com> References: <4AB36E49.2030102@barcelonamedia.org> <4AB383AD.2060704@gmail.com> Message-ID: <4AB39A50.9010707@barcelonamedia.org> Thanks for the answer! > I think you need to interleave several mono channels from audiotestsrc > into a stereo stream and then shove them into the vorbis encoder. You > need an audio mixer, or something like that. > I would like to encode separate channels. For example, this runs gst-launch-0.10 -v interleave name=i ! queue ! \ vorbisenc ! vorbisdec ! \ jackaudiosink connect=none \ jackaudiosrc ! audioconvert ! queue ! i. \ jackaudiosrc ! audioconvert ! queue ! i. but this does not: gst-launch-0.10 -v interleave name=i ! queue ! \ vorbisenc ! vorbisdec ! \ jackaudiosink connect=none \ jackaudiosrc ! audioconvert ! queue ! i. \ jackaudiosrc ! audioconvert ! queue ! i. \ jackaudiosrc ! audioconvert ! queue ! i. \ jackaudiosrc ! audioconvert ! queue ! i. Do you know why? The vorbis spec allows for 255 channels and I simply would like to run n channels through the vorbis encoder ... I really could use some help. Thanks in advance, Dirk From tristan at sat.qc.ca Fri Sep 18 17:30:19 2009 From: tristan at sat.qc.ca (Tristan Matthews) Date: Fri, 18 Sep 2009 11:30:19 -0400 Subject: [gst-devel] multichannel vorbis In-Reply-To: <4AB39A50.9010707@barcelonamedia.org> References: <4AB36E49.2030102@barcelonamedia.org> <4AB383AD.2060704@gmail.com> <4AB39A50.9010707@barcelonamedia.org> Message-ID: <4AB3A78B.5000104@sat.qc.ca> You probably have to set the "channel-positions" property on the interleave element, which you can't do with gst-launch as I recall (i.e. you need to write c or python app for it) as it is an array. However for more than 8 channels the positions might have to all be set to GST_AUDIO_CHANNEL_POSITION_NONE Are you just trying to do something like this? gst-launch -v jackaudiosrc connect=none ! audio/x-raw-float, channels=24 ! vorbisenc ! vorbisdec ! jackaudiosink connect=none (note that the bottleneck here will probably be your soundcard). -Tristan Dirk Griffioen wrote: > Thanks for the answer! > >> I think you need to interleave several mono channels from audiotestsrc >> into a stereo stream and then shove them into the vorbis encoder. You >> need an audio mixer, or something like that. >> >> > I would like to encode separate channels. > > For example, this runs > > gst-launch-0.10 -v interleave name=i ! queue ! \ > vorbisenc ! vorbisdec ! \ > jackaudiosink connect=none \ > jackaudiosrc ! audioconvert ! queue ! i. \ > jackaudiosrc ! audioconvert ! queue ! i. > > but this does not: > > gst-launch-0.10 -v interleave name=i ! queue ! \ > vorbisenc ! vorbisdec ! \ > jackaudiosink connect=none \ > jackaudiosrc ! audioconvert ! queue ! i. \ > jackaudiosrc ! audioconvert ! queue ! i. \ > jackaudiosrc ! audioconvert ! queue ! i. \ > jackaudiosrc ! audioconvert ! queue ! i. > > Do you know why? The vorbis spec allows for 255 channels and I simply > would like to run n channels through the vorbis encoder ... > > I really could use some help. > > Thanks in advance, Dirk > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Tristan Matthews Soci?t? des arts technologiques [SAT] email: tristan at sat.qc.ca web: http://www.music.mcgill.ca/~tmatthews From dirk.griffioen at barcelonamedia.org Fri Sep 18 17:59:32 2009 From: dirk.griffioen at barcelonamedia.org (Dirk Griffioen) Date: Fri, 18 Sep 2009 17:59:32 +0200 Subject: [gst-devel] multichannel vorbis In-Reply-To: <4AB3A78B.5000104@sat.qc.ca> References: <4AB36E49.2030102@barcelonamedia.org> <4AB383AD.2060704@gmail.com> <4AB39A50.9010707@barcelonamedia.org> <4AB3A78B.5000104@sat.qc.ca> Message-ID: <4AB3AE64.9060602@barcelonamedia.org> Hi Tristan, > You probably have to set the "channel-positions" property on the > interleave element, which you can't do with gst-launch as I recall (i.e. > you need to write c or python app for it) as it is an array. However for > more than 8 channels the positions might have to all be set to > GST_AUDIO_CHANNEL_POSITION_NONE > > Yes I read about that (http://tristanswork.blogspot.com/2008/08/multichannel-audio-with-gstreamer.html), and can I do this in python as well? (I dont mind C, but python will be quicker). > Are you just trying to do something like this? > > gst-launch -v jackaudiosrc connect=none ! audio/x-raw-float, channels=24 > ! vorbisenc ! vorbisdec ! jackaudiosink connect=none > > Well, almost :) I am trying to put rtp in between the vorbisencoder and decoder so I can stream n channels from A to B over a single rtp session I get the following on the receiving end (after copying the new config string from A to B): GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:sink: caps = audio/x-vorbis WARNING: from element /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: Could not decode stream. Additional debug info: vorbisdec.c(670): vorbis_handle_identification_packet (): /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: Using NONE channel layout for more than 8 channels Which is weird because it knows this: /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:src: caps = audio/x-raw-float, rate=(int)48000, channels=(int)24, endianness=(int)1234, width=(int)32, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE > Do you have any tips? (Maybe the NONE layout is not in the vorbis config string ...) Regards, Dirk PS - how do you call the 'audio/x-raw-float, channels=24' element? > (note that the bottleneck here will probably be your soundcard). > > -Tristan > > Dirk Griffioen wrote: > >> Thanks for the answer! >> >> >>> I think you need to interleave several mono channels from audiotestsrc >>> into a stereo stream and then shove them into the vorbis encoder. You >>> need an audio mixer, or something like that. >>> >>> >>> >> I would like to encode separate channels. >> >> For example, this runs >> >> gst-launch-0.10 -v interleave name=i ! queue ! \ >> vorbisenc ! vorbisdec ! \ >> jackaudiosink connect=none \ >> jackaudiosrc ! audioconvert ! queue ! i. \ >> jackaudiosrc ! audioconvert ! queue ! i. >> >> but this does not: >> >> gst-launch-0.10 -v interleave name=i ! queue ! \ >> vorbisenc ! vorbisdec ! \ >> jackaudiosink connect=none \ >> jackaudiosrc ! audioconvert ! queue ! i. \ >> jackaudiosrc ! audioconvert ! queue ! i. \ >> jackaudiosrc ! audioconvert ! queue ! i. \ >> jackaudiosrc ! audioconvert ! queue ! i. >> >> Do you know why? The vorbis spec allows for 255 channels and I simply >> would like to run n channels through the vorbis encoder ... >> >> I really could use some help. >> >> Thanks in advance, Dirk >> >> >> ------------------------------------------------------------------------------ >> Come build with us! The BlackBerry® Developer Conference in SF, CA >> is the only developer event you need to attend this year. Jumpstart your >> developing skills, take BlackBerry mobile applications to market and stay >> ahead of the curve. Join us from November 9-12, 2009. Register now! >> http://p.sf.net/sfu/devconf >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From tristan at sat.qc.ca Fri Sep 18 19:20:02 2009 From: tristan at sat.qc.ca (Tristan Matthews) Date: Fri, 18 Sep 2009 13:20:02 -0400 Subject: [gst-devel] multichannel vorbis In-Reply-To: <4AB3AE64.9060602@barcelonamedia.org> References: <4AB36E49.2030102@barcelonamedia.org> <4AB383AD.2060704@gmail.com> <4AB39A50.9010707@barcelonamedia.org> <4AB3A78B.5000104@sat.qc.ca> <4AB3AE64.9060602@barcelonamedia.org> Message-ID: <4AB3C142.5090400@sat.qc.ca> As far as I know this should be fine in python, though I haven't tried it. Our app (https://svn.sat.qc.ca/trac/miville) does this for up to 8 channels of vorbis or raw audio (with rtpL16pay/depay) and gstrtpbin. We haven't tried (yet) to implement support for more than 8 channels. Here we don't set the channel positions and it works, but I do get that same "warning could not decode stream" even though the sound if fine. The element to set the number of channels is a caps filter element, so the equivalent in C would be: GstElement *capsfilter; gst_element_factory_make(capsfilter, NULL); g_object_set(capsfilter, "caps", "audio/x-raw-float, channels=8", NULL); and then link it in between jackaudiosrc and vorbisenc. -Tristan Dirk Griffioen wrote: > Hi Tristan, >> You probably have to set the "channel-positions" property on the >> interleave element, which you can't do with gst-launch as I recall (i.e. >> you need to write c or python app for it) as it is an array. However for >> more than 8 channels the positions might have to all be set to >> GST_AUDIO_CHANNEL_POSITION_NONE >> >> > Yes I read about that > (http://tristanswork.blogspot.com/2008/08/multichannel-audio-with-gstreamer.html), > and can I do this in python as well? (I dont mind C, but python will > be quicker). > >> Are you just trying to do something like this? >> >> gst-launch -v jackaudiosrc connect=none ! audio/x-raw-float, channels=24 >> ! vorbisenc ! vorbisdec ! jackaudiosink connect=none >> >> > Well, almost :) > > I am trying to put rtp in between the vorbisencoder and decoder so I > can stream n channels from A to B over a single rtp session > > I get the following on the receiving end (after copying the new config > string from A to B): > > GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:sink: caps = > audio/x-vorbis > WARNING: from element /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: > Could not decode stream. > Additional debug info: > vorbisdec.c(670): vorbis_handle_identification_packet (): > /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: > Using NONE channel layout for more than 8 channels > > Which is weird because it knows this: > > /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:src: caps = > audio/x-raw-float, rate=(int)48000, channels=(int)24, > endianness=(int)1234, width=(int)32, > channel-positions=(GstAudioChannelPosition)< > GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, > GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, > GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, > GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, > GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, > GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, > GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, > GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, > GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, > GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, > GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, > GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE > > > Do you have any tips? (Maybe the NONE layout is not in the vorbis > config string ...) > > Regards, Dirk > > PS - how do you call the 'audio/x-raw-float, channels=24' element? > >> (note that the bottleneck here will probably be your soundcard). >> >> -Tristan >> >> Dirk Griffioen wrote: >> >>> Thanks for the answer! >>> >>> >>>> I think you need to interleave several mono channels from audiotestsrc >>>> into a stereo stream and then shove them into the vorbis encoder. You >>>> need an audio mixer, or something like that. >>>> >>>> >>>> >>> I would like to encode separate channels. >>> >>> For example, this runs >>> >>> gst-launch-0.10 -v interleave name=i ! queue ! \ >>> vorbisenc ! vorbisdec ! \ >>> jackaudiosink connect=none \ >>> jackaudiosrc ! audioconvert ! queue ! i. \ >>> jackaudiosrc ! audioconvert ! queue ! i. >>> >>> but this does not: >>> >>> gst-launch-0.10 -v interleave name=i ! queue ! \ >>> vorbisenc ! vorbisdec ! \ >>> jackaudiosink connect=none \ >>> jackaudiosrc ! audioconvert ! queue ! i. \ >>> jackaudiosrc ! audioconvert ! queue ! i. \ >>> jackaudiosrc ! audioconvert ! queue ! i. \ >>> jackaudiosrc ! audioconvert ! queue ! i. >>> >>> Do you know why? The vorbis spec allows for 255 channels and I simply >>> would like to run n channels through the vorbis encoder ... >>> >>> I really could use some help. >>> >>> Thanks in advance, Dirk >>> >>> >>> ------------------------------------------------------------------------------ >>> Come build with us! The BlackBerry® Developer Conference in SF, CA >>> is the only developer event you need to attend this year. Jumpstart your >>> developing skills, take BlackBerry mobile applications to market and stay >>> ahead of the curve. Join us from November 9-12, 2009. Register now! >>> http://p.sf.net/sfu/devconf >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >>> >> >> >> > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Tristan Matthews Soci?t? des arts technologiques [SAT] email: tristan at sat.qc.ca web: http://www.music.mcgill.ca/~tmatthews From tristan at sat.qc.ca Fri Sep 18 19:29:27 2009 From: tristan at sat.qc.ca (Tristan Matthews) Date: Fri, 18 Sep 2009 13:29:27 -0400 Subject: [gst-devel] multichannel vorbis In-Reply-To: <4AB3C142.5090400@sat.qc.ca> References: <4AB36E49.2030102@barcelonamedia.org> <4AB383AD.2060704@gmail.com> <4AB39A50.9010707@barcelonamedia.org> <4AB3A78B.5000104@sat.qc.ca> <4AB3AE64.9060602@barcelonamedia.org> <4AB3C142.5090400@sat.qc.ca> Message-ID: <4AB3C377.3050606@sat.qc.ca> This example might help: sender gst-launch -v jackaudiosrc connect=none ! audio/x-raw-float, channels=10 ! vorbisenc ! rtpvorbispay ! udpsink port=10000 receiver gst-launch -v udpsrc caps="$CAPS" port=10000 ! rtpvorbisdepay ! vorbisdec ! queue max-size-buffers=3 ! jackaudiosink connect=none where $CAPS are the caps of the udpsink from the first pipeline. -Tristan Tristan Matthews wrote: > As far as I know this should be fine in python, though I haven't tried > it. Our app (https://svn.sat.qc.ca/trac/miville) does this for up to 8 > channels of vorbis or raw audio (with rtpL16pay/depay) and gstrtpbin. We > haven't tried (yet) to implement support for more than 8 channels. Here > we don't set the channel positions and it works, but I do get that same > "warning could not decode stream" even though the sound if fine. > The element to set the number of channels is a caps filter element, so > the equivalent in C would be: > > GstElement *capsfilter; > gst_element_factory_make(capsfilter, NULL); > g_object_set(capsfilter, "caps", "audio/x-raw-float, channels=8", NULL); > > and then link it in between jackaudiosrc and vorbisenc. > > -Tristan > > Dirk Griffioen wrote: > >> Hi Tristan, >> >>> You probably have to set the "channel-positions" property on the >>> interleave element, which you can't do with gst-launch as I recall (i.e. >>> you need to write c or python app for it) as it is an array. However for >>> more than 8 channels the positions might have to all be set to >>> GST_AUDIO_CHANNEL_POSITION_NONE >>> >>> >>> >> Yes I read about that >> (http://tristanswork.blogspot.com/2008/08/multichannel-audio-with-gstreamer.html), >> and can I do this in python as well? (I dont mind C, but python will >> be quicker). >> >> >>> Are you just trying to do something like this? >>> >>> gst-launch -v jackaudiosrc connect=none ! audio/x-raw-float, channels=24 >>> ! vorbisenc ! vorbisdec ! jackaudiosink connect=none >>> >>> >>> >> Well, almost :) >> >> I am trying to put rtp in between the vorbisencoder and decoder so I >> can stream n channels from A to B over a single rtp session >> >> I get the following on the receiving end (after copying the new config >> string from A to B): >> >> GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:sink: caps = >> audio/x-vorbis >> WARNING: from element /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: >> Could not decode stream. >> Additional debug info: >> vorbisdec.c(670): vorbis_handle_identification_packet (): >> /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: >> Using NONE channel layout for more than 8 channels >> >> Which is weird because it knows this: >> >> /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:src: caps = >> audio/x-raw-float, rate=(int)48000, channels=(int)24, >> endianness=(int)1234, width=(int)32, >> channel-positions=(GstAudioChannelPosition)< >> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE > >> >> Do you have any tips? (Maybe the NONE layout is not in the vorbis >> config string ...) >> >> Regards, Dirk >> >> PS - how do you call the 'audio/x-raw-float, channels=24' element? >> >> >>> (note that the bottleneck here will probably be your soundcard). >>> >>> -Tristan >>> >>> Dirk Griffioen wrote: >>> >>> >>>> Thanks for the answer! >>>> >>>> >>>> >>>>> I think you need to interleave several mono channels from audiotestsrc >>>>> into a stereo stream and then shove them into the vorbis encoder. You >>>>> need an audio mixer, or something like that. >>>>> >>>>> >>>>> >>>>> >>>> I would like to encode separate channels. >>>> >>>> For example, this runs >>>> >>>> gst-launch-0.10 -v interleave name=i ! queue ! \ >>>> vorbisenc ! vorbisdec ! \ >>>> jackaudiosink connect=none \ >>>> jackaudiosrc ! audioconvert ! queue ! i. \ >>>> jackaudiosrc ! audioconvert ! queue ! i. >>>> >>>> but this does not: >>>> >>>> gst-launch-0.10 -v interleave name=i ! queue ! \ >>>> vorbisenc ! vorbisdec ! \ >>>> jackaudiosink connect=none \ >>>> jackaudiosrc ! audioconvert ! queue ! i. \ >>>> jackaudiosrc ! audioconvert ! queue ! i. \ >>>> jackaudiosrc ! audioconvert ! queue ! i. \ >>>> jackaudiosrc ! audioconvert ! queue ! i. >>>> >>>> Do you know why? The vorbis spec allows for 255 channels and I simply >>>> would like to run n channels through the vorbis encoder ... >>>> >>>> I really could use some help. >>>> >>>> Thanks in advance, Dirk >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Come build with us! The BlackBerry® Developer Conference in SF, CA >>>> is the only developer event you need to attend this year. Jumpstart your >>>> developing skills, take BlackBerry mobile applications to market and stay >>>> ahead of the curve. Join us from November 9-12, 2009. Register now! >>>> http://p.sf.net/sfu/devconf >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> gstreamer-devel at lists.sourceforge.net >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>>> >>>> >>>> >>> >>> >> ------------------------------------------------------------------------ >> >> ------------------------------------------------------------------------------ >> Come build with us! The BlackBerry® Developer Conference in SF, CA >> is the only developer event you need to attend this year. Jumpstart your >> developing skills, take BlackBerry mobile applications to market and stay >> ahead of the curve. Join us from November 9-12, 2009. Register now! >> http://p.sf.net/sfu/devconf >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > -- Tristan Matthews Soci?t? des arts technologiques [SAT] email: tristan at sat.qc.ca web: http://www.music.mcgill.ca/~tmatthews From daltoncezane at gmail.com Fri Sep 18 21:42:43 2009 From: daltoncezane at gmail.com (=?ISO-8859-1?Q?Dalton_C=E9zane?=) Date: Fri, 18 Sep 2009 16:42:43 -0300 Subject: [gst-devel] Demux H264 video and HE-AAC audio Message-ID: <98029fd50909181242v33c613dem2e0d68f3905d1d77@mail.gmail.com> Hi all, I am new at list and at the GStreamer study. I am trying to demux H264 video and HE-AAC audio with gst-launch-0.10. I already succeeded just the video, without audio, with this command line: gst-launch-0.10 filesrc location="GLOBO188bytes.mpg" ! ffdemux_mpegts name=demux demux. queue ! ffdec_h264 ! xvimagesink ! demux. queue ! faad ! audioconvert ! osssink This way, the video is displayed but the sound does not play. Can anyone help me? Some tip? Thanks in advance. -- ======================================================= Dalton C?zane - Voip UFCG: 1075-2005 Mestrando em Ci?ncia da Computa??o (UFCG) Bacharel em Ci?ncia da Computa??o (UFCG) T?cnico em Inform?tica (ETER) -------------- next part -------------- An HTML attachment was scrubbed... URL: From sledgehammer_999 at hotmail.com Sat Sep 19 01:44:20 2009 From: sledgehammer_999 at hotmail.com (sledge hammer) Date: Sat, 19 Sep 2009 02:44:20 +0300 Subject: [gst-devel] Demux H264 video and HE-AAC audio In-Reply-To: <98029fd50909181242v33c613dem2e0d68f3905d1d77@mail.gmail.com> References: <98029fd50909181242v33c613dem2e0d68f3905d1d77@mail.gmail.com> Message-ID: maybe try 'mpegpsdemux' instead of 'ffdemux_mpegts' Date: Fri, 18 Sep 2009 16:42:43 -0300 From: daltoncezane at gmail.com To: gstreamer-devel at lists.sourceforge.net Subject: [gst-devel] Demux H264 video and HE-AAC audio Hi all, I am new at list and at the GStreamer study. I am trying to demux H264 video and HE-AAC audio with gst-launch-0.10. I already succeeded just the video, without audio, with this command line: gst-launch-0.10 filesrc location="GLOBO188bytes.mpg" ! ffdemux_mpegts name=demux demux. queue ! ffdec_h264 ! xvimagesink ! demux. queue ! faad ! audioconvert ! osssink This way, the video is displayed but the sound does not play. Can anyone help me? Some tip? Thanks in advance. -- ======================================================= Dalton C?zane - Voip UFCG: 1075-2005 Mestrando em Ci?ncia da Computa??o (UFCG) Bacharel em Ci?ncia da Computa??o (UFCG) T?cnico em Inform?tica (ETER) _________________________________________________________________ ????? Messenger; ????? ??? Windows Live. ?????? ???????????. http://microsoft.com/windows/windowslive -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilboed at gmail.com Sat Sep 19 11:10:44 2009 From: bilboed at gmail.com (Edward Hervey) Date: Sat, 19 Sep 2009 11:10:44 +0200 Subject: [gst-devel] Demux H264 video and HE-AAC audio In-Reply-To: <98029fd50909181242v33c613dem2e0d68f3905d1d77@mail.gmail.com> References: <98029fd50909181242v33c613dem2e0d68f3905d1d77@mail.gmail.com> Message-ID: <1253351444.8398.0.camel@putamadre> Use mpegtsdemux and not ffdemux_mpegts. The rule of thumb is : don't use the ffmpeg demuxers (they have a rank of NONE for a reason). Edward On Fri, 2009-09-18 at 16:42 -0300, Dalton C?zane wrote: > Hi all, > I am new at list and at the GStreamer study. I am trying to demux H264 > video and HE-AAC audio with gst-launch-0.10. > I already succeeded just the video, without audio, with this command > line: gst-launch-0.10 filesrc location="GLOBO188bytes.mpg" ! > ffdemux_mpegts name=demux demux. queue ! ffdec_h264 ! xvimagesink ! > demux. queue ! faad ! audioconvert ! osssink > > This way, the video is displayed but the sound does not play. > Can anyone help me? Some tip? > > Thanks in advance. > > -- > ======================================================= > Dalton C?zane - Voip UFCG: 1075-2005 > Mestrando em Ci?ncia da Computa??o (UFCG) > Bacharel em Ci?ncia da Computa??o (UFCG) > T?cnico em Inform?tica (ETER) > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From smcnam at gmail.com Sat Sep 19 17:47:00 2009 From: smcnam at gmail.com (Sean McNamara) Date: Sat, 19 Sep 2009 11:47:00 -0400 Subject: [gst-devel] Demux H264 video and HE-AAC audio In-Reply-To: <98029fd50909181242v33c613dem2e0d68f3905d1d77@mail.gmail.com> References: <98029fd50909181242v33c613dem2e0d68f3905d1d77@mail.gmail.com> Message-ID: <74eb1fe20909190847r4f1c8414m95e9bd3cec251b1b@mail.gmail.com> Are you sure the problem isn't with osssink? Post the entire output of using gst-launch with this command, and you may also try autoaudiosink instead of osssink. If you're sure the problem isn't with the audiosink, I'll need to see the rest of the output to know what exactly went wrong. HTH, Sean On 9/18/09, Dalton C?zane wrote: > Hi all, > I am new at list and at the GStreamer study. I am trying to demux H264 video > and HE-AAC audio with gst-launch-0.10. > I already succeeded just the video, without audio, with this command line: > gst-launch-0.10 filesrc location="GLOBO188bytes.mpg" ! ffdemux_mpegts > name=demux demux. queue ! ffdec_h264 ! xvimagesink ! demux. queue ! faad ! > audioconvert ! osssink > > This way, the video is displayed but the sound does not play. > Can anyone help me? Some tip? > > Thanks in advance. > > -- > ======================================================= > Dalton C?zane - Voip UFCG: 1075-2005 > Mestrando em Ci?ncia da Computa??o (UFCG) > Bacharel em Ci?ncia da Computa??o (UFCG) > T?cnico em Inform?tica (ETER) > From fthiery at gmail.com Sat Sep 19 21:18:45 2009 From: fthiery at gmail.com (Florent) Date: Sat, 19 Sep 2009 21:18:45 +0200 Subject: [gst-devel] Find out what sinks are avaiable. In-Reply-To: <4AA67795.90905@hora-obscura.de> References: <25254429.post@talk.nabble.com> <4AA67795.90905@hora-obscura.de> Message-ID: <1efe3a6e0909191218t5c9005a3m906fe0857556e2f8@mail.gmail.com> See /proc/asound/cards file contents for details on the devices that are available on your system. Florent From mmikkone at mail.student.oulu.fi Sat Sep 19 21:54:31 2009 From: mmikkone at mail.student.oulu.fi (Marko Mikkonen) Date: Sat, 19 Sep 2009 22:54:31 +0300 Subject: [gst-devel] SDLVideoSink hangs on Windows References: <008553576B5D4B998823E3526BC7102A@casapascaa><4A8C34F1.9010100@gmail.com><369EECFB9942443093F2CFDF11C74E3B@casapascaa><772db3280908211333j1aff23f5k3fdf1bbec9690de6@mail.gmail.com><772db3280908211340v1d1ef6b9t1ca73f375c58d008@mail.gmail.com> Message-ID: <498D494F5A9E4FA29ADDA85DAE29EBF3@casapascaa> I've compiled now GStreamer with MinGW for Windows, but I can't get SDLVideoSink to work. It hangs: -------------------- C:\TEMP\gstreamer test>gst-launch-0.10.exe filesrc location="H:/movies/test.avi" ! decodebin ! sdlvideosink --gst-plugin-path="c:\temp\gstreamer test" Setting pipeline to PAUSED ... Pipeline is PREROLLING ... ------------------ After this nothing happens. Gst-launch doesn't respond to ctrl-C and I have to kill the process in Task Manager. SDLAudioSink works fine. BTW I've compiled SDLSink plugin with both configure/make/make install and with CodeBlocks MinGW edition (created a new project for it). I got a lot less rubbish in the dll with CodeBlocks. I mean there was a lot of exported functions in the dll when I compiled it with Make. Compiling with CodeBlocks made only 5 functions exported. How do I debug this? I'm completely new to debugging GStreamer. I tried to set breakpoints with CodeBlocks, but gdb is very slow to even catch the first breakpoint so I gave that up (has something to do with loading all those plugins in plugin directory???). I tried printing from SDLVideoSink - functions are only called once when making gstreamer registry (when I've deleted it) so I'm not becoming any wiser. Then I tried to use --gst-debug option with some debug categories, but I'm not really sure which ones to try. There are so many and there's a lot of output... Maybe you could point me to the right direction? Could this hanging be caused by a race condition somewhere (there's no eternal loop here: CPU usage stays minimal)? -Marko From dirk.griffioen at barcelonamedia.org Mon Sep 21 12:03:51 2009 From: dirk.griffioen at barcelonamedia.org (Dirk Griffioen) Date: Mon, 21 Sep 2009 12:03:51 +0200 Subject: [gst-devel] multichannel vorbis In-Reply-To: <4AB3C377.3050606@sat.qc.ca> References: <4AB36E49.2030102@barcelonamedia.org> <4AB383AD.2060704@gmail.com> <4AB39A50.9010707@barcelonamedia.org> <4AB3A78B.5000104@sat.qc.ca> <4AB3AE64.9060602@barcelonamedia.org> <4AB3C142.5090400@sat.qc.ca> <4AB3C377.3050606@sat.qc.ca> Message-ID: <4AB74F87.8090508@barcelonamedia.org> Hi Tristan, Thanks for the replies. They are really helpfull! (And I will have a further look at 'miville' - it looks really nice). > This example might help: > sender > > gst-launch -v jackaudiosrc connect=none ! audio/x-raw-float, channels=10 > ! vorbisenc ! rtpvorbispay ! udpsink port=10000 > > receiver > > gst-launch -v udpsrc caps="$CAPS" port=10000 ! rtpvorbisdepay ! > vorbisdec ! queue max-size-buffers=3 ! jackaudiosink connect=none > > where $CAPS are the caps of the udpsink from the first pipeline. > > This does not work for me, jackaudiosink does not pop up in qjackctl ... I tried some other configurations, but nothing. However, somehow my first pipeline with rtp decided to work, with 24 channels and from gst-launch. Still, I get: WARNING: from element /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: Could not decode stream. Additional debug info: vorbisdec.c(670): vorbis_handle_identification_packet (): /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: Using NONE channel layout for more than 8 channels Maybe this can interpreted as 'cannot read layout from stream, defaulting to NONE' - as the audio streams fine. > -Tristan > > Tristan Matthews wrote: > >> As far as I know this should be fine in python, though I haven't tried >> it. Our app (https://svn.sat.qc.ca/trac/miville) does this for up to 8 >> channels of vorbis or raw audio (with rtpL16pay/depay) and gstrtpbin. We >> haven't tried (yet) to implement support for more than 8 channels. Here >> we don't set the channel positions and it works, but I do get that same >> "warning could not decode stream" even though the sound if fine. >> The element to set the number of channels is a caps filter element, so >> the equivalent in C would be: >> >> GstElement *capsfilter; >> gst_element_factory_make(capsfilter, NULL); >> g_object_set(capsfilter, "caps", "audio/x-raw-float, channels=8", NULL); >> >> and then link it in between jackaudiosrc and vorbisenc. >> >> -Tristan >> >> Dirk Griffioen wrote: >> >> >>> Hi Tristan, >>> >>> >>>> You probably have to set the "channel-positions" property on the >>>> interleave element, which you can't do with gst-launch as I recall (i.e. >>>> you need to write c or python app for it) as it is an array. However for >>>> more than 8 channels the positions might have to all be set to >>>> GST_AUDIO_CHANNEL_POSITION_NONE >>>> >>>> >>>> >>>> >>> Yes I read about that >>> (http://tristanswork.blogspot.com/2008/08/multichannel-audio-with-gstreamer.html), >>> and can I do this in python as well? (I dont mind C, but python will >>> be quicker). >>> >>> >>> >>>> Are you just trying to do something like this? >>>> >>>> gst-launch -v jackaudiosrc connect=none ! audio/x-raw-float, channels=24 >>>> ! vorbisenc ! vorbisdec ! jackaudiosink connect=none >>>> >>>> >>>> >>>> >>> Well, almost :) >>> >>> I am trying to put rtp in between the vorbisencoder and decoder so I >>> can stream n channels from A to B over a single rtp session >>> >>> I get the following on the receiving end (after copying the new config >>> string from A to B): >>> >>> GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:sink: caps = >>> audio/x-vorbis >>> WARNING: from element /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: >>> Could not decode stream. >>> Additional debug info: >>> vorbisdec.c(670): vorbis_handle_identification_packet (): >>> /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: >>> Using NONE channel layout for more than 8 channels >>> >>> Which is weird because it knows this: >>> >>> /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:src: caps = >>> audio/x-raw-float, rate=(int)48000, channels=(int)24, >>> endianness=(int)1234, width=(int)32, >>> channel-positions=(GstAudioChannelPosition)< >>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE > >>> >>> Do you have any tips? (Maybe the NONE layout is not in the vorbis >>> config string ...) >>> >>> Regards, Dirk >>> >>> PS - how do you call the 'audio/x-raw-float, channels=24' element? >>> >>> >>> >>>> (note that the bottleneck here will probably be your soundcard). >>>> >>>> -Tristan >>>> >>>> Dirk Griffioen wrote: >>>> >>>> >>>> >>>>> Thanks for the answer! >>>>> >>>>> >>>>> >>>>> >>>>>> I think you need to interleave several mono channels from audiotestsrc >>>>>> into a stereo stream and then shove them into the vorbis encoder. You >>>>>> need an audio mixer, or something like that. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> I would like to encode separate channels. >>>>> >>>>> For example, this runs >>>>> >>>>> gst-launch-0.10 -v interleave name=i ! queue ! \ >>>>> vorbisenc ! vorbisdec ! \ >>>>> jackaudiosink connect=none \ >>>>> jackaudiosrc ! audioconvert ! queue ! i. \ >>>>> jackaudiosrc ! audioconvert ! queue ! i. >>>>> >>>>> but this does not: >>>>> >>>>> gst-launch-0.10 -v interleave name=i ! queue ! \ >>>>> vorbisenc ! vorbisdec ! \ >>>>> jackaudiosink connect=none \ >>>>> jackaudiosrc ! audioconvert ! queue ! i. \ >>>>> jackaudiosrc ! audioconvert ! queue ! i. \ >>>>> jackaudiosrc ! audioconvert ! queue ! i. \ >>>>> jackaudiosrc ! audioconvert ! queue ! i. >>>>> >>>>> Do you know why? The vorbis spec allows for 255 channels and I simply >>>>> would like to run n channels through the vorbis encoder ... >>>>> >>>>> I really could use some help. >>>>> >>>>> Thanks in advance, Dirk >>>>> >>>>> >>>>> ------------------------------------------------------------------------------ >>>>> Come build with us! The BlackBerry® Developer Conference in SF, CA >>>>> is the only developer event you need to attend this year. Jumpstart your >>>>> developing skills, take BlackBerry mobile applications to market and stay >>>>> ahead of the curve. Join us from November 9-12, 2009. Register now! >>>>> http://p.sf.net/sfu/devconf >>>>> _______________________________________________ >>>>> gstreamer-devel mailing list >>>>> gstreamer-devel at lists.sourceforge.net >>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>> >>>>> >>>>> >>>>> >>>>> >>>> >>>> >>>> >>> ------------------------------------------------------------------------ >>> >>> ------------------------------------------------------------------------------ >>> Come build with us! The BlackBerry® Developer Conference in SF, CA >>> is the only developer event you need to attend this year. Jumpstart your >>> developing skills, take BlackBerry mobile applications to market and stay >>> ahead of the curve. Join us from November 9-12, 2009. Register now! >>> http://p.sf.net/sfu/devconf >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >>> >> >> > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From tristan at sat.qc.ca Mon Sep 21 14:54:13 2009 From: tristan at sat.qc.ca (Tristan Matthews) Date: Mon, 21 Sep 2009 08:54:13 -0400 Subject: [gst-devel] multichannel vorbis In-Reply-To: <4AB74F87.8090508@barcelonamedia.org> References: <4AB36E49.2030102@barcelonamedia.org> <4AB383AD.2060704@gmail.com> <4AB39A50.9010707@barcelonamedia.org> <4AB3A78B.5000104@sat.qc.ca> <4AB3AE64.9060602@barcelonamedia.org> <4AB3C142.5090400@sat.qc.ca> <4AB3C377.3050606@sat.qc.ca> <4AB74F87.8090508@barcelonamedia.org> Message-ID: <4AB77775.5040408@sat.qc.ca> Are you copying the caps from udpsink's sink pad directly from the sender pipeline? Vorbis caps (i.e. the codebook) will change for different configurations. -T Dirk Griffioen wrote: > Hi Tristan, > > Thanks for the replies. They are really helpfull! (And I will have a > further look at 'miville' - it looks really nice). >> This example might help: >> sender >> >> gst-launch -v jackaudiosrc connect=none ! audio/x-raw-float, channels=10 >> ! vorbisenc ! rtpvorbispay ! udpsink port=10000 >> >> receiver >> >> gst-launch -v udpsrc caps="$CAPS" port=10000 ! rtpvorbisdepay ! >> vorbisdec ! queue max-size-buffers=3 ! jackaudiosink connect=none >> >> where $CAPS are the caps of the udpsink from the first pipeline. >> >> > This does not work for me, jackaudiosink does not pop up in qjackctl > ... I tried some other configurations, but nothing. > > However, somehow my first pipeline with rtp decided to work, with 24 > channels and from gst-launch. Still, I get: > > WARNING: from element /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: > Could not decode stream. > Additional debug info: > vorbisdec.c(670): vorbis_handle_identification_packet (): > /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: > Using NONE channel layout for more than 8 channels > > Maybe this can interpreted as 'cannot read layout from stream, > defaulting to NONE' - as the audio streams fine. >> -Tristan >> >> Tristan Matthews wrote: >> >>> As far as I know this should be fine in python, though I haven't tried >>> it. Our app (https://svn.sat.qc.ca/trac/miville) does this for up to 8 >>> channels of vorbis or raw audio (with rtpL16pay/depay) and gstrtpbin. We >>> haven't tried (yet) to implement support for more than 8 channels. Here >>> we don't set the channel positions and it works, but I do get that same >>> "warning could not decode stream" even though the sound if fine. >>> The element to set the number of channels is a caps filter element, so >>> the equivalent in C would be: >>> >>> GstElement *capsfilter; >>> gst_element_factory_make(capsfilter, NULL); >>> g_object_set(capsfilter, "caps", "audio/x-raw-float, channels=8", NULL); >>> >>> and then link it in between jackaudiosrc and vorbisenc. >>> >>> -Tristan >>> >>> Dirk Griffioen wrote: >>> >>> >>>> Hi Tristan, >>>> >>>> >>>>> You probably have to set the "channel-positions" property on the >>>>> interleave element, which you can't do with gst-launch as I recall (i.e. >>>>> you need to write c or python app for it) as it is an array. However for >>>>> more than 8 channels the positions might have to all be set to >>>>> GST_AUDIO_CHANNEL_POSITION_NONE >>>>> >>>>> >>>>> >>>>> >>>> Yes I read about that >>>> (http://tristanswork.blogspot.com/2008/08/multichannel-audio-with-gstreamer.html), >>>> and can I do this in python as well? (I dont mind C, but python will >>>> be quicker). >>>> >>>> >>>> >>>>> Are you just trying to do something like this? >>>>> >>>>> gst-launch -v jackaudiosrc connect=none ! audio/x-raw-float, channels=24 >>>>> ! vorbisenc ! vorbisdec ! jackaudiosink connect=none >>>>> >>>>> >>>>> >>>>> >>>> Well, almost :) >>>> >>>> I am trying to put rtp in between the vorbisencoder and decoder so I >>>> can stream n channels from A to B over a single rtp session >>>> >>>> I get the following on the receiving end (after copying the new config >>>> string from A to B): >>>> >>>> GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:sink: caps = >>>> audio/x-vorbis >>>> WARNING: from element /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: >>>> Could not decode stream. >>>> Additional debug info: >>>> vorbisdec.c(670): vorbis_handle_identification_packet (): >>>> /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: >>>> Using NONE channel layout for more than 8 channels >>>> >>>> Which is weird because it knows this: >>>> >>>> /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:src: caps = >>>> audio/x-raw-float, rate=(int)48000, channels=(int)24, >>>> endianness=(int)1234, width=(int)32, >>>> channel-positions=(GstAudioChannelPosition)< >>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE > >>>> >>>> Do you have any tips? (Maybe the NONE layout is not in the vorbis >>>> config string ...) >>>> >>>> Regards, Dirk >>>> >>>> PS - how do you call the 'audio/x-raw-float, channels=24' element? >>>> >>>> >>>> >>>>> (note that the bottleneck here will probably be your soundcard). >>>>> >>>>> -Tristan >>>>> >>>>> Dirk Griffioen wrote: >>>>> >>>>> >>>>> >>>>>> Thanks for the answer! >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> I think you need to interleave several mono channels from audiotestsrc >>>>>>> into a stereo stream and then shove them into the vorbis encoder. You >>>>>>> need an audio mixer, or something like that. >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> I would like to encode separate channels. >>>>>> >>>>>> For example, this runs >>>>>> >>>>>> gst-launch-0.10 -v interleave name=i ! queue ! \ >>>>>> vorbisenc ! vorbisdec ! \ >>>>>> jackaudiosink connect=none \ >>>>>> jackaudiosrc ! audioconvert ! queue ! i. \ >>>>>> jackaudiosrc ! audioconvert ! queue ! i. >>>>>> >>>>>> but this does not: >>>>>> >>>>>> gst-launch-0.10 -v interleave name=i ! queue ! \ >>>>>> vorbisenc ! vorbisdec ! \ >>>>>> jackaudiosink connect=none \ >>>>>> jackaudiosrc ! audioconvert ! queue ! i. \ >>>>>> jackaudiosrc ! audioconvert ! queue ! i. \ >>>>>> jackaudiosrc ! audioconvert ! queue ! i. \ >>>>>> jackaudiosrc ! audioconvert ! queue ! i. >>>>>> >>>>>> Do you know why? The vorbis spec allows for 255 channels and I simply >>>>>> would like to run n channels through the vorbis encoder ... >>>>>> >>>>>> I really could use some help. >>>>>> >>>>>> Thanks in advance, Dirk >>>>>> >>>>>> >>>>>> ------------------------------------------------------------------------------ >>>>>> Come build with us! The BlackBerry® Developer Conference in SF, CA >>>>>> is the only developer event you need to attend this year. Jumpstart your >>>>>> developing skills, take BlackBerry mobile applications to market and stay >>>>>> ahead of the curve. Join us from November 9-12, 2009. Register now! >>>>>> http://p.sf.net/sfu/devconf >>>>>> _______________________________________________ >>>>>> gstreamer-devel mailing list >>>>>> gstreamer-devel at lists.sourceforge.net >>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>> ------------------------------------------------------------------------ >>>> >>>> ------------------------------------------------------------------------------ >>>> Come build with us! The BlackBerry® Developer Conference in SF, CA >>>> is the only developer event you need to attend this year. Jumpstart your >>>> developing skills, take BlackBerry mobile applications to market and stay >>>> ahead of the curve. Join us from November 9-12, 2009. Register now! >>>> http://p.sf.net/sfu/devconf >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> gstreamer-devel mailing list >>>> gstreamer-devel at lists.sourceforge.net >>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>> >>>> >>>> >>> >>> >> >> >> > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Tristan Matthews Soci?t? des arts technologiques [SAT] email: tristan at sat.qc.ca web: http://www.music.mcgill.ca/~tmatthews From julien.isorce at gmail.com Mon Sep 21 14:15:02 2009 From: julien.isorce at gmail.com (Julien Isorce) Date: Mon, 21 Sep 2009 14:15:02 +0200 Subject: [gst-devel] UDPSRC BUG In-Reply-To: <3c1737210909150959y2d51346bibf060169902dc9c@mail.gmail.com> References: <881487A9950B044AA6861B700374AAAF02C3C012@deimsg40.de.net.world> <180a127d0909150852mca47c9ela0b1325589aa512b@mail.gmail.com> <3c1737210909150959y2d51346bibf060169902dc9c@mail.gmail.com> Message-ID: <180a127d0909210515n48db935ah8b9c38c803861eb5@mail.gmail.com> 2009/9/15 Michael Smith > > I believe this patch makes multiudpsink work on windows: > https://bugzilla.gnome.org/show_bug.cgi?id=534243 > Hi, The patch does not apply to the current git. Sincerely Julien > > Note that I'm still waiting for more windows testing (particularly on > vista), and hopefully a code review from someone who knows this stuff. > > udpsrc might also have similar issues, I'm not sure - I can fix these > if people are willing to test the code and comment on that bug. > > Mike > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From dirk.griffioen at barcelonamedia.org Mon Sep 21 15:10:33 2009 From: dirk.griffioen at barcelonamedia.org (Dirk Griffioen) Date: Mon, 21 Sep 2009 15:10:33 +0200 Subject: [gst-devel] multichannel vorbis In-Reply-To: <4AB77775.5040408@sat.qc.ca> References: <4AB36E49.2030102@barcelonamedia.org> <4AB383AD.2060704@gmail.com> <4AB39A50.9010707@barcelonamedia.org> <4AB3A78B.5000104@sat.qc.ca> <4AB3AE64.9060602@barcelonamedia.org> <4AB3C142.5090400@sat.qc.ca> <4AB3C377.3050606@sat.qc.ca> <4AB74F87.8090508@barcelonamedia.org> <4AB77775.5040408@sat.qc.ca> Message-ID: <4AB77B49.3030900@barcelonamedia.org> Tristan., > Are you copying the caps from udpsink's sink pad directly from the > sender pipeline? Vorbis caps (i.e. the codebook) will change for > different configurations. > > That is what I did: both ends appear to be running (the commandline just runs); the receiver end says: Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstSystemClock But the jackaudiosink does not show up in qjackctl - nor is there any indication or other output why it fails ... Best, Dirk > -T > > Dirk Griffioen wrote: > >> Hi Tristan, >> >> Thanks for the replies. They are really helpfull! (And I will have a >> further look at 'miville' - it looks really nice). >> >>> This example might help: >>> sender >>> >>> gst-launch -v jackaudiosrc connect=none ! audio/x-raw-float, channels=10 >>> ! vorbisenc ! rtpvorbispay ! udpsink port=10000 >>> >>> receiver >>> >>> gst-launch -v udpsrc caps="$CAPS" port=10000 ! rtpvorbisdepay ! >>> vorbisdec ! queue max-size-buffers=3 ! jackaudiosink connect=none >>> >>> where $CAPS are the caps of the udpsink from the first pipeline. >>> >>> >>> >> This does not work for me, jackaudiosink does not pop up in qjackctl >> ... I tried some other configurations, but nothing. >> >> However, somehow my first pipeline with rtp decided to work, with 24 >> channels and from gst-launch. Still, I get: >> >> WARNING: from element /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: >> Could not decode stream. >> Additional debug info: >> vorbisdec.c(670): vorbis_handle_identification_packet (): >> /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: >> Using NONE channel layout for more than 8 channels >> >> Maybe this can interpreted as 'cannot read layout from stream, >> defaulting to NONE' - as the audio streams fine. >> >>> -Tristan >>> >>> Tristan Matthews wrote: >>> >>> >>>> As far as I know this should be fine in python, though I haven't tried >>>> it. Our app (https://svn.sat.qc.ca/trac/miville) does this for up to 8 >>>> channels of vorbis or raw audio (with rtpL16pay/depay) and gstrtpbin. We >>>> haven't tried (yet) to implement support for more than 8 channels. Here >>>> we don't set the channel positions and it works, but I do get that same >>>> "warning could not decode stream" even though the sound if fine. >>>> The element to set the number of channels is a caps filter element, so >>>> the equivalent in C would be: >>>> >>>> GstElement *capsfilter; >>>> gst_element_factory_make(capsfilter, NULL); >>>> g_object_set(capsfilter, "caps", "audio/x-raw-float, channels=8", NULL); >>>> >>>> and then link it in between jackaudiosrc and vorbisenc. >>>> >>>> -Tristan >>>> >>>> Dirk Griffioen wrote: >>>> >>>> >>>> >>>>> Hi Tristan, >>>>> >>>>> >>>>> >>>>>> You probably have to set the "channel-positions" property on the >>>>>> interleave element, which you can't do with gst-launch as I recall (i.e. >>>>>> you need to write c or python app for it) as it is an array. However for >>>>>> more than 8 channels the positions might have to all be set to >>>>>> GST_AUDIO_CHANNEL_POSITION_NONE >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> Yes I read about that >>>>> (http://tristanswork.blogspot.com/2008/08/multichannel-audio-with-gstreamer.html), >>>>> and can I do this in python as well? (I dont mind C, but python will >>>>> be quicker). >>>>> >>>>> >>>>> >>>>> >>>>>> Are you just trying to do something like this? >>>>>> >>>>>> gst-launch -v jackaudiosrc connect=none ! audio/x-raw-float, channels=24 >>>>>> ! vorbisenc ! vorbisdec ! jackaudiosink connect=none >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> Well, almost :) >>>>> >>>>> I am trying to put rtp in between the vorbisencoder and decoder so I >>>>> can stream n channels from A to B over a single rtp session >>>>> >>>>> I get the following on the receiving end (after copying the new config >>>>> string from A to B): >>>>> >>>>> GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:sink: caps = >>>>> audio/x-vorbis >>>>> WARNING: from element /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: >>>>> Could not decode stream. >>>>> Additional debug info: >>>>> vorbisdec.c(670): vorbis_handle_identification_packet (): >>>>> /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: >>>>> Using NONE channel layout for more than 8 channels >>>>> >>>>> Which is weird because it knows this: >>>>> >>>>> /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:src: caps = >>>>> audio/x-raw-float, rate=(int)48000, channels=(int)24, >>>>> endianness=(int)1234, width=(int)32, >>>>> channel-positions=(GstAudioChannelPosition)< >>>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE > >>>>> >>>>> Do you have any tips? (Maybe the NONE layout is not in the vorbis >>>>> config string ...) >>>>> >>>>> Regards, Dirk >>>>> >>>>> PS - how do you call the 'audio/x-raw-float, channels=24' element? >>>>> >>>>> >>>>> >>>>> >>>>>> (note that the bottleneck here will probably be your soundcard). >>>>>> >>>>>> -Tristan >>>>>> >>>>>> Dirk Griffioen wrote: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> Thanks for the answer! >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>>> I think you need to interleave several mono channels from audiotestsrc >>>>>>>> into a stereo stream and then shove them into the vorbis encoder. You >>>>>>>> need an audio mixer, or something like that. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> I would like to encode separate channels. >>>>>>> >>>>>>> For example, this runs >>>>>>> >>>>>>> gst-launch-0.10 -v interleave name=i ! queue ! \ >>>>>>> vorbisenc ! vorbisdec ! \ >>>>>>> jackaudiosink connect=none \ >>>>>>> jackaudiosrc ! audioconvert ! queue ! i. \ >>>>>>> jackaudiosrc ! audioconvert ! queue ! i. >>>>>>> >>>>>>> but this does not: >>>>>>> >>>>>>> gst-launch-0.10 -v interleave name=i ! queue ! \ >>>>>>> vorbisenc ! vorbisdec ! \ >>>>>>> jackaudiosink connect=none \ >>>>>>> jackaudiosrc ! audioconvert ! queue ! i. \ >>>>>>> jackaudiosrc ! audioconvert ! queue ! i. \ >>>>>>> jackaudiosrc ! audioconvert ! queue ! i. \ >>>>>>> jackaudiosrc ! audioconvert ! queue ! i. >>>>>>> >>>>>>> Do you know why? The vorbis spec allows for 255 channels and I simply >>>>>>> would like to run n channels through the vorbis encoder ... >>>>>>> >>>>>>> I really could use some help. >>>>>>> >>>>>>> Thanks in advance, Dirk >>>>>>> >>>>>>> >>>>>>> ------------------------------------------------------------------------------ >>>>>>> Come build with us! The BlackBerry® Developer Conference in SF, CA >>>>>>> is the only developer event you need to attend this year. Jumpstart your >>>>>>> developing skills, take BlackBerry mobile applications to market and stay >>>>>>> ahead of the curve. Join us from November 9-12, 2009. Register now! >>>>>>> http://p.sf.net/sfu/devconf >>>>>>> _______________________________________________ >>>>>>> gstreamer-devel mailing list >>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>> ------------------------------------------------------------------------ >>>>> >>>>> ------------------------------------------------------------------------------ >>>>> Come build with us! The BlackBerry® Developer Conference in SF, CA >>>>> is the only developer event you need to attend this year. Jumpstart your >>>>> developing skills, take BlackBerry mobile applications to market and stay >>>>> ahead of the curve. Join us from November 9-12, 2009. Register now! >>>>> http://p.sf.net/sfu/devconf >>>>> ------------------------------------------------------------------------ >>>>> >>>>> _______________________________________________ >>>>> gstreamer-devel mailing list >>>>> gstreamer-devel at lists.sourceforge.net >>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>> >>>>> >>>>> >>>>> >>>> >>> >>> >> ------------------------------------------------------------------------ >> >> ------------------------------------------------------------------------------ >> Come build with us! The BlackBerry® Developer Conference in SF, CA >> is the only developer event you need to attend this year. Jumpstart your >> developing skills, take BlackBerry mobile applications to market and stay >> ahead of the curve. Join us from November 9-12, 2009. Register now! >> http://p.sf.net/sfu/devconf >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > -- > Tristan Matthews > Soci?t? des arts technologiques [SAT] > email: tristan at sat.qc.ca > web: http://www.music.mcgill.ca/~tmatthews > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From dirk.griffioen at barcelonamedia.org Mon Sep 21 15:46:35 2009 From: dirk.griffioen at barcelonamedia.org (Dirk Griffioen) Date: Mon, 21 Sep 2009 15:46:35 +0200 Subject: [gst-devel] multichannel vorbis In-Reply-To: <4AB77775.5040408@sat.qc.ca> References: <4AB36E49.2030102@barcelonamedia.org> <4AB383AD.2060704@gmail.com> <4AB39A50.9010707@barcelonamedia.org> <4AB3A78B.5000104@sat.qc.ca> <4AB3AE64.9060602@barcelonamedia.org> <4AB3C142.5090400@sat.qc.ca> <4AB3C377.3050606@sat.qc.ca> <4AB74F87.8090508@barcelonamedia.org> <4AB77775.5040408@sat.qc.ca> Message-ID: <4AB783BB.1060101@barcelonamedia.org> Tristan, If I add 'host' to the udpsink it works ... gst-launch -v jackaudiosrc connect=none ! audio/x-raw-float, channels=10 ! vorbisenc ! rtpvorbispay ! udpsink port=10000 host=127.0.0.1 Then both jackaudiosink and jackaudiosrc show up. Thanks for the help! Best, Dirk > Are you copying the caps from udpsink's sink pad directly from the > sender pipeline? Vorbis caps (i.e. the codebook) will change for > different configurations. > > -T > > Dirk Griffioen wrote: > >> Hi Tristan, >> >> Thanks for the replies. They are really helpfull! (And I will have a >> further look at 'miville' - it looks really nice). >> >>> This example might help: >>> sender >>> >>> gst-launch -v jackaudiosrc connect=none ! audio/x-raw-float, channels=10 >>> ! vorbisenc ! rtpvorbispay ! udpsink port=10000 >>> >>> receiver >>> >>> gst-launch -v udpsrc caps="$CAPS" port=10000 ! rtpvorbisdepay ! >>> vorbisdec ! queue max-size-buffers=3 ! jackaudiosink connect=none >>> >>> where $CAPS are the caps of the udpsink from the first pipeline. >>> >>> >>> >> This does not work for me, jackaudiosink does not pop up in qjackctl >> ... I tried some other configurations, but nothing. >> >> However, somehow my first pipeline with rtp decided to work, with 24 >> channels and from gst-launch. Still, I get: >> >> WARNING: from element /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: >> Could not decode stream. >> Additional debug info: >> vorbisdec.c(670): vorbis_handle_identification_packet (): >> /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: >> Using NONE channel layout for more than 8 channels >> >> Maybe this can interpreted as 'cannot read layout from stream, >> defaulting to NONE' - as the audio streams fine. >> >>> -Tristan >>> >>> Tristan Matthews wrote: >>> >>> >>>> As far as I know this should be fine in python, though I haven't tried >>>> it. Our app (https://svn.sat.qc.ca/trac/miville) does this for up to 8 >>>> channels of vorbis or raw audio (with rtpL16pay/depay) and gstrtpbin. We >>>> haven't tried (yet) to implement support for more than 8 channels. Here >>>> we don't set the channel positions and it works, but I do get that same >>>> "warning could not decode stream" even though the sound if fine. >>>> The element to set the number of channels is a caps filter element, so >>>> the equivalent in C would be: >>>> >>>> GstElement *capsfilter; >>>> gst_element_factory_make(capsfilter, NULL); >>>> g_object_set(capsfilter, "caps", "audio/x-raw-float, channels=8", NULL); >>>> >>>> and then link it in between jackaudiosrc and vorbisenc. >>>> >>>> -Tristan >>>> >>>> Dirk Griffioen wrote: >>>> >>>> >>>> >>>>> Hi Tristan, >>>>> >>>>> >>>>> >>>>>> You probably have to set the "channel-positions" property on the >>>>>> interleave element, which you can't do with gst-launch as I recall (i.e. >>>>>> you need to write c or python app for it) as it is an array. However for >>>>>> more than 8 channels the positions might have to all be set to >>>>>> GST_AUDIO_CHANNEL_POSITION_NONE >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> Yes I read about that >>>>> (http://tristanswork.blogspot.com/2008/08/multichannel-audio-with-gstreamer.html), >>>>> and can I do this in python as well? (I dont mind C, but python will >>>>> be quicker). >>>>> >>>>> >>>>> >>>>> >>>>>> Are you just trying to do something like this? >>>>>> >>>>>> gst-launch -v jackaudiosrc connect=none ! audio/x-raw-float, channels=24 >>>>>> ! vorbisenc ! vorbisdec ! jackaudiosink connect=none >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> Well, almost :) >>>>> >>>>> I am trying to put rtp in between the vorbisencoder and decoder so I >>>>> can stream n channels from A to B over a single rtp session >>>>> >>>>> I get the following on the receiving end (after copying the new config >>>>> string from A to B): >>>>> >>>>> GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:sink: caps = >>>>> audio/x-vorbis >>>>> WARNING: from element /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: >>>>> Could not decode stream. >>>>> Additional debug info: >>>>> vorbisdec.c(670): vorbis_handle_identification_packet (): >>>>> /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: >>>>> Using NONE channel layout for more than 8 channels >>>>> >>>>> Which is weird because it knows this: >>>>> >>>>> /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:src: caps = >>>>> audio/x-raw-float, rate=(int)48000, channels=(int)24, >>>>> endianness=(int)1234, width=(int)32, >>>>> channel-positions=(GstAudioChannelPosition)< >>>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, >>>>> GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE > >>>>> >>>>> Do you have any tips? (Maybe the NONE layout is not in the vorbis >>>>> config string ...) >>>>> >>>>> Regards, Dirk >>>>> >>>>> PS - how do you call the 'audio/x-raw-float, channels=24' element? >>>>> >>>>> >>>>> >>>>> >>>>>> (note that the bottleneck here will probably be your soundcard). >>>>>> >>>>>> -Tristan >>>>>> >>>>>> Dirk Griffioen wrote: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> Thanks for the answer! >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>>> I think you need to interleave several mono channels from audiotestsrc >>>>>>>> into a stereo stream and then shove them into the vorbis encoder. You >>>>>>>> need an audio mixer, or something like that. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> I would like to encode separate channels. >>>>>>> >>>>>>> For example, this runs >>>>>>> >>>>>>> gst-launch-0.10 -v interleave name=i ! queue ! \ >>>>>>> vorbisenc ! vorbisdec ! \ >>>>>>> jackaudiosink connect=none \ >>>>>>> jackaudiosrc ! audioconvert ! queue ! i. \ >>>>>>> jackaudiosrc ! audioconvert ! queue ! i. >>>>>>> >>>>>>> but this does not: >>>>>>> >>>>>>> gst-launch-0.10 -v interleave name=i ! queue ! \ >>>>>>> vorbisenc ! vorbisdec ! \ >>>>>>> jackaudiosink connect=none \ >>>>>>> jackaudiosrc ! audioconvert ! queue ! i. \ >>>>>>> jackaudiosrc ! audioconvert ! queue ! i. \ >>>>>>> jackaudiosrc ! audioconvert ! queue ! i. \ >>>>>>> jackaudiosrc ! audioconvert ! queue ! i. >>>>>>> >>>>>>> Do you know why? The vorbis spec allows for 255 channels and I simply >>>>>>> would like to run n channels through the vorbis encoder ... >>>>>>> >>>>>>> I really could use some help. >>>>>>> >>>>>>> Thanks in advance, Dirk >>>>>>> >>>>>>> >>>>>>> ------------------------------------------------------------------------------ >>>>>>> Come build with us! The BlackBerry® Developer Conference in SF, CA >>>>>>> is the only developer event you need to attend this year. Jumpstart your >>>>>>> developing skills, take BlackBerry mobile applications to market and stay >>>>>>> ahead of the curve. Join us from November 9-12, 2009. Register now! >>>>>>> http://p.sf.net/sfu/devconf >>>>>>> _______________________________________________ >>>>>>> gstreamer-devel mailing list >>>>>>> gstreamer-devel at lists.sourceforge.net >>>>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>> ------------------------------------------------------------------------ >>>>> >>>>> ------------------------------------------------------------------------------ >>>>> Come build with us! The BlackBerry® Developer Conference in SF, CA >>>>> is the only developer event you need to attend this year. Jumpstart your >>>>> developing skills, take BlackBerry mobile applications to market and stay >>>>> ahead of the curve. Join us from November 9-12, 2009. Register now! >>>>> http://p.sf.net/sfu/devconf >>>>> ------------------------------------------------------------------------ >>>>> >>>>> _______________________________________________ >>>>> gstreamer-devel mailing list >>>>> gstreamer-devel at lists.sourceforge.net >>>>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>>>> >>>>> >>>>> >>>>> >>>> >>> >>> >> ------------------------------------------------------------------------ >> >> ------------------------------------------------------------------------------ >> Come build with us! The BlackBerry® Developer Conference in SF, CA >> is the only developer event you need to attend this year. Jumpstart your >> developing skills, take BlackBerry mobile applications to market and stay >> ahead of the curve. Join us from November 9-12, 2009. Register now! >> http://p.sf.net/sfu/devconf >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > -- > Tristan Matthews > Soci?t? des arts technologiques [SAT] > email: tristan at sat.qc.ca > web: http://www.music.mcgill.ca/~tmatthews > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From tristan at sat.qc.ca Mon Sep 21 17:22:43 2009 From: tristan at sat.qc.ca (Tristan Matthews) Date: Mon, 21 Sep 2009 11:22:43 -0400 Subject: [gst-devel] multichannel vorbis In-Reply-To: <4AB783BB.1060101@barcelonamedia.org> References: <4AB36E49.2030102@barcelonamedia.org> <4AB383AD.2060704@gmail.com> <4AB39A50.9010707@barcelonamedia.org> <4AB3A78B.5000104@sat.qc.ca> <4AB3AE64.9060602@barcelonamedia.org> <4AB3C142.5090400@sat.qc.ca> <4AB3C377.3050606@sat.qc.ca> <4AB74F87.8090508@barcelonamedia.org> <4AB77775.5040408@sat.qc.ca> <4AB783BB.1060101@barcelonamedia.org> Message-ID: <1c7708560909210822m2a285b06g912eaedba5a937b4@mail.gmail.com> Interesting, I had noticed this discrepancy between ubuntu 9.04 and 8.04 (i'm on 8.04 on my work machine) before. For some reason, "localhost" (the default) doesn't work on 9.04, but 127.0.0.1 does. Thanks for confirming, I just filed this under: https://bugzilla.gnome.org/show_bug.cgi?id=595840 -T 2009/9/21 Dirk Griffioen > Tristan, > > If I add 'host' to the udpsink it works ... > > gst-launch -v jackaudiosrc connect=none ! audio/x-raw-float, channels=10 > ! vorbisenc ! rtpvorbispay ! udpsink port=10000 host=127.0.0.1 > > > Then both jackaudiosink and jackaudiosrc show up. > > Thanks for the help! > > Best, Dirk > > Are you copying the caps from udpsink's sink pad directly from the > sender pipeline? Vorbis caps (i.e. the codebook) will change for > different configurations. > > -T > > Dirk Griffioen wrote: > > > Hi Tristan, > > Thanks for the replies. They are really helpfull! (And I will have a > further look at 'miville' - it looks really nice). > > > This example might help: > sender > > gst-launch -v jackaudiosrc connect=none ! audio/x-raw-float, channels=10 > ! vorbisenc ! rtpvorbispay ! udpsink port=10000 > > receiver > > gst-launch -v udpsrc caps="$CAPS" port=10000 ! rtpvorbisdepay ! > vorbisdec ! queue max-size-buffers=3 ! jackaudiosink connect=none > > where $CAPS are the caps of the udpsink from the first pipeline. > > > > > This does not work for me, jackaudiosink does not pop up in qjackctl > ... I tried some other configurations, but nothing. > > However, somehow my first pipeline with rtp decided to work, with 24 > channels and from gst-launch. Still, I get: > > WARNING: from element /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: > Could not decode stream. > Additional debug info: > vorbisdec.c(670): vorbis_handle_identification_packet (): > /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: > Using NONE channel layout for more than 8 channels > > Maybe this can interpreted as 'cannot read layout from stream, > defaulting to NONE' - as the audio streams fine. > > > -Tristan > > Tristan Matthews wrote: > > > > As far as I know this should be fine in python, though I haven't tried > it. Our app (https://svn.sat.qc.ca/trac/miville) does this for up to 8 > channels of vorbis or raw audio (with rtpL16pay/depay) and gstrtpbin. We > haven't tried (yet) to implement support for more than 8 channels. Here > we don't set the channel positions and it works, but I do get that same > "warning could not decode stream" even though the sound if fine. > The element to set the number of channels is a caps filter element, so > the equivalent in C would be: > > GstElement *capsfilter; > gst_element_factory_make(capsfilter, NULL); > g_object_set(capsfilter, "caps", "audio/x-raw-float, channels=8", NULL); > > and then link it in between jackaudiosrc and vorbisenc. > > -Tristan > > Dirk Griffioen wrote: > > > > > Hi Tristan, > > > > > You probably have to set the "channel-positions" property on the > interleave element, which you can't do with gst-launch as I recall (i.e. > you need to write c or python app for it) as it is an array. However for > more than 8 channels the positions might have to all be set to > GST_AUDIO_CHANNEL_POSITION_NONE > > > > > > > Yes I read about that > (http://tristanswork.blogspot.com/2008/08/multichannel-audio-with-gstreamer.html), > and can I do this in python as well? (I dont mind C, but python will > be quicker). > > > > > > Are you just trying to do something like this? > > gst-launch -v jackaudiosrc connect=none ! audio/x-raw-float, channels=24 > ! vorbisenc ! vorbisdec ! jackaudiosink connect=none > > > > > > > Well, almost :) > > I am trying to put rtp in between the vorbisencoder and decoder so I > can stream n channels from A to B over a single rtp session > > I get the following on the receiving end (after copying the new config > string from A to B): > > GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:sink: caps = > audio/x-vorbis > WARNING: from element /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: > Could not decode stream. > Additional debug info: > vorbisdec.c(670): vorbis_handle_identification_packet (): > /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0: > Using NONE channel layout for more than 8 channels > > Which is weird because it knows this: > > /GstPipeline:pipeline0/GstVorbisDec:vorbisdec0.GstPad:src: caps = > audio/x-raw-float, rate=(int)48000, channels=(int)24, > endianness=(int)1234, width=(int)32, > channel-positions=(GstAudioChannelPosition)< > GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, > GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, > GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, > GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, > GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, > GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, > GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, > GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, > GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, > GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, > GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE, > GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE > > > Do you have any tips? (Maybe the NONE layout is not in the vorbis > config string ...) > > Regards, Dirk > > PS - how do you call the 'audio/x-raw-float, channels=24' element? > > > > > > (note that the bottleneck here will probably be your soundcard). > > -Tristan > > Dirk Griffioen wrote: > > > > > > Thanks for the answer! > > > > > > > I think you need to interleave several mono channels from audiotestsrc > into a stereo stream and then shove them into the vorbis encoder. You > need an audio mixer, or something like that. > > > > > > > > I would like to encode separate channels. > > For example, this runs > > gst-launch-0.10 -v interleave name=i ! queue ! \ > vorbisenc ! vorbisdec ! \ > jackaudiosink connect=none \ > jackaudiosrc ! audioconvert ! queue ! i. \ > jackaudiosrc ! audioconvert ! queue ! i. > > but this does not: > > gst-launch-0.10 -v interleave name=i ! queue ! \ > vorbisenc ! vorbisdec ! \ > jackaudiosink connect=none \ > jackaudiosrc ! audioconvert ! queue ! i. \ > jackaudiosrc ! audioconvert ! queue ! i. \ > jackaudiosrc ! audioconvert ! queue ! i. \ > jackaudiosrc ! audioconvert ! queue ! i. > > Do you know why? The vorbis spec allows for 255 channels and I simply > would like to run n channels through the vorbis encoder ... > > I really could use some help. > > Thanks in advance, Dirk > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now!http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing listgstreamer-devel at lists.sourceforge.nethttps://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now!http://p.sf.net/sfu/devconf > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing listgstreamer-devel at lists.sourceforge.nethttps://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now!http://p.sf.net/sfu/devconf > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing listgstreamer-devel at lists.sourceforge.nethttps://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- > Tristan Matthews > Soci?t? des arts technologiques [SAT] > email: tristan at sat.qc.ca > web: http://www.music.mcgill.ca/~tmatthews > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now!http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing listgstreamer-devel at lists.sourceforge.nethttps://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- Tristan Matthews email: tristan at sat.qc.ca web: http://tristanswork.blogspot.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From confusosk8 at gmail.com Tue Sep 22 00:02:33 2009 From: confusosk8 at gmail.com (Gabriel Duarte) Date: Mon, 21 Sep 2009 19:02:33 -0300 Subject: [gst-devel] Slow capture on dv1394src Message-ID: <669348040909211502r3500c2ic972c4e287101460@mail.gmail.com> Hello all! This pipeline works well, but the frame rate is very slow, any idea how to get it better? *gst-launch-0.10 dv1394src ! queue ! dvdemux ! ffdec_dvvideo ! ffmpegcolorspace ! video/x-raw-yuv, width=720, height=480, bpp=32, depth=24 ! autovideosink* Thanks ;) gst-launch-0.10 dv1394src ! queue ! dvdemux ! ffdec_dvvideo ! ffmpegcolorspace ! video/x-raw-yuv, width=720, height=480, bpp=32, depth=24 ! autovideosink -- Gabriel Duarte Linux User #471185 Rio de Janeiro - RJ http://kinuxlinux.org/gabriel_duarte Phones: (55) (21) 9463-7760 /*Mobile*/ (55) (21) 2464-9302 /*Home*/ (55) (21) 2529-5080 /*Work*/ -----BEGIN GEEK CODE BLOCK----- Version: 3.12 GCS d- s: a--- C++ UL+++ P L++++ E- W+ N++ o++ K++ w--- O- M- V- PS++ PE++ Y PGP- t++ 5-- X+++ R tv++ b++ DI+ D++ G++ e+ h* r+ y++++ ------END GEEK CODE BLOCK------ -------------- next part -------------- An HTML attachment was scrubbed... URL: From dirk.griffioen at barcelonamedia.org Tue Sep 22 16:08:40 2009 From: dirk.griffioen at barcelonamedia.org (Dirk Griffioen) Date: Tue, 22 Sep 2009 16:08:40 +0200 Subject: [gst-devel] on-bye-ssrc not implemented? Message-ID: <4AB8DA68.2020607@barcelonamedia.org> Hi, I have a small python-gst app based on http://blog.abourget.net/2009/6/14/gstreamer-rtp-and-live-streaming. This works great! However, when I implement a callback for the 'on-bye-ssrc' like def on_bye_ssrc(self, session, ssrc, userdata): print 'bye', session, ssrc self.rtpbin.connect('on-new-ssrc', self.on_new_ssrc) it is never called when the rtp producer leaves the scene (killing it or stopping the producing pipeline programatically by calling udpsink_rtcpout.set_locked_state(gst.STATE_NULL)). I believe it should be called as rfc 3550 states ' A site sends the RTCP BYE packet (Section 6.6) when it leaves the conference. What am I doing wrong? As on-new-ssrc does get called ... Thanks in advance! Dirk From dirk.griffioen at barcelonamedia.org Tue Sep 22 17:06:26 2009 From: dirk.griffioen at barcelonamedia.org (Dirk Griffioen) Date: Tue, 22 Sep 2009 17:06:26 +0200 Subject: [gst-devel] on-bye-ssrc not implemented? In-Reply-To: <4AB8DA68.2020607@barcelonamedia.org> References: <4AB8DA68.2020607@barcelonamedia.org> Message-ID: <4AB8E7F2.6040902@barcelonamedia.org> Sorry, wrong example. I meant: def on_bye_ssrc(self, session, ssrc, userdata): print 'bye', session, ssrc rtpbin.connect('on-bye-ssrc', on_bye_ssrc) Best, Dirk > Hi, > > I have a small python-gst app based on > http://blog.abourget.net/2009/6/14/gstreamer-rtp-and-live-streaming. > > This works great! > > However, when I implement a callback for the 'on-bye-ssrc' like > > def on_bye_ssrc(self, session, ssrc, userdata): > print 'bye', session, ssrc > > self.rtpbin.connect('on-new-ssrc', self.on_new_ssrc) > > it is never called when the rtp producer leaves the scene (killing it or > stopping the producing pipeline programatically by calling > udpsink_rtcpout.set_locked_state(gst.STATE_NULL)). > > I believe it should be called as rfc 3550 states ' > > A site sends the RTCP BYE packet (Section 6.6) when it leaves the conference. > > > What am I doing wrong? As on-new-ssrc does get called ... > > Thanks in advance! > > Dirk > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From olivier.aubert at liris.cnrs.fr Tue Sep 22 17:29:12 2009 From: olivier.aubert at liris.cnrs.fr (Olivier Aubert) Date: Tue, 22 Sep 2009 17:29:12 +0200 Subject: [gst-devel] New SVGOverlay element Message-ID: <1253633352.5022.40.camel@pennylane> Hello I have implemented a new SVGOverlay element (attached), in python, using the example buffer-draw.py as codebase. It appropriately works for my own needs (although I have a strange locking problem in one of my pipeline). I post it here in case it might be of use for someone else (and for coding comments also). As my application is python-based, it is enough for my needs. However, if there is interest (and if the approach is validated), I could translate this in C for inclusion as a standard gst element. Regards, Olivier -------------- next part -------------- A non-text attachment was scrubbed... Name: svgoverlay.py Type: text/x-python Size: 5926 bytes Desc: not available URL: From lists at svrinformatica.it Tue Sep 22 19:12:49 2009 From: lists at svrinformatica.it (Mailing List SVR) Date: Tue, 22 Sep 2009 19:12:49 +0200 Subject: [gst-devel] udpsrc,multiudpsink and fd not closed Message-ID: <200909221912.49593.lists@svrinformatica.it> Hi all, I wrote a very simple gstreamer app: - I receive rtp/rtsp stream using udpsrc and rtspsrc and forward them with multiudpsink, all works quite well but the number od open files increase over the time and seems no one release (attached is lsof output). What is the right way to release fd and not increase them over the time? I read about closefd property but I not set it and the default is true, any hints? thanks Nicola -------------- next part -------------- lsof | grep python python 25080 root cwd DIR 8,4 4096 1472819 /usr/local/server python 25080 root rtd DIR 8,4 4096 2 / python 25080 root txt REG 8,4 2268568 1399705 /usr/bin/python2.6 python 25080 root mem REG 8,4 184236 1455459 /usr/lib/gstreamer-0.10/libgstcoreelements.so python 25080 root mem REG 8,4 192480 1457106 /usr/lib/gstreamer-0.10/libgstrtpmanager.so python 25080 root mem REG 8,4 9544 1399389 /usr/lib/libgstnetbuffer-0.10.so.0.18.0 python 25080 root mem REG 8,4 55404 1457304 /usr/lib/gstreamer-0.10/libgstudp.so python 25080 root mem REG 8,4 141276 1399381 /usr/lib/libgstaudio-0.10.so.0.18.0 python 25080 root mem REG 8,4 63480 1399384 /usr/lib/libgsttag-0.10.so.0.18.0 python 25080 root mem REG 8,4 50548 1399392 /usr/lib/libgstriff-0.10.so.0.18.0 python 25080 root mem REG 8,4 117252 1457278 /usr/lib/gstreamer-0.10/libgstasf.so python 25080 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1401876 pipe python 25080 root 31w FIFO 0,6 1401876 pipe python 25080 root 32r FIFO 0,6 1401903 pipe python 25080 root 33w FIFO 0,6 1401903 pipe python 25080 root 34u FIFO 0,6 1752802 pipe python 25080 root 35u FIFO 0,6 1761153 pipe python 25080 root 36u IPv6 1761148 UDP *:57407 python 25080 root 37u IPv6 1403970 UDP *:35350 python 25080 root 38r FIFO 0,6 1401915 pipe python 25080 root 39w FIFO 0,6 1401915 pipe python 25080 root 40r FIFO 0,6 1403968 pipe python 25080 root 41w FIFO 0,6 1403968 pipe python 25080 root 42u unix 0xf59ab340 1403971 socket python 25080 root 43u unix 0xf59aba40 1403972 socket python 25080 root 44u IPv4 1403976 TCP 192.168.2.220:46679->192.168.2.17:rtsp (ESTABLISHED) python 25080 root 45r FIFO 0,6 1404041 pipe python 25080 root 46u IPv4 1403983 UDP *:49872 python 25080 root 47u unix 0xf5986e00 1403984 socket python 25080 root 48u unix 0xf59868c0 1403985 socket python 25080 root 49u IPv4 1403987 UDP *:49873 python 25080 root 50u unix 0xf5986a80 1403988 socket python 25080 root 51u unix 0xf5987340 1403989 socket python 25080 root 52u IPv4 1404003 UDP *:40604 python 25080 root 53u unix 0xf5987dc0 1404004 socket python 25080 root 54u unix 0xf5986fc0 1404005 socket python 25080 root 55u IPv4 1404007 UDP *:40605 python 25080 root 56u unix 0xf59876c0 1404008 socket python 25080 root 57u unix 0xf5987880 1404009 socket python 25080 root 58w FIFO 0,6 1404041 pipe python 25080 root 59r FIFO 0,6 1415438 pipe python 25080 root 60r FIFO 0,6 1411323 pipe python 25080 root 61w FIFO 0,6 1411323 pipe python 25080 root 62r FIFO 0,6 1411329 pipe python 25080 root 63r FIFO 0,6 1404061 pipe python 25080 root 64w FIFO 0,6 1404061 pipe python 25080 root 65r FIFO 0,6 1404129 pipe python 25080 root 66w FIFO 0,6 1411329 pipe python 25080 root 67r FIFO 0,6 1412353 pipe python 25080 root 68w FIFO 0,6 1412353 pipe python 25080 root 69r FIFO 0,6 1415415 pipe python 25080 root 70w FIFO 0,6 1415415 pipe python 25080 root 71w FIFO 0,6 1415438 pipe python 25080 root 72r FIFO 0,6 1415453 pipe python 25080 root 73w FIFO 0,6 1415453 pipe python 25080 root 74r FIFO 0,6 1415466 pipe python 25080 root 75w FIFO 0,6 1415466 pipe python 25080 root 76r FIFO 0,6 1418606 pipe python 25080 root 77r FIFO 0,6 1417540 pipe python 25080 root 78w FIFO 0,6 1404129 pipe python 25080 root 79r FIFO 0,6 1404135 pipe python 25080 root 80w FIFO 0,6 1404135 pipe python 25080 root 81r FIFO 0,6 1404141 pipe python 25080 root 82w FIFO 0,6 1404141 pipe python 25080 root 83w FIFO 0,6 1417540 pipe python 25080 root 84r FIFO 0,6 1409242 pipe python 25080 root 85w FIFO 0,6 1409242 pipe python 25080 root 86w FIFO 0,6 1418606 pipe python 25080 root 87r FIFO 0,6 1419629 pipe python 25080 root 88w FIFO 0,6 1419629 pipe python 25080 root 89r FIFO 0,6 1428099 pipe python 25080 root 90r FIFO 0,6 1428118 pipe python 25080 root 91w FIFO 0,6 1428118 pipe python 25080 root 92r FIFO 0,6 1428123 pipe python 25080 root 93w FIFO 0,6 1428123 pipe python 25080 root 94r FIFO 0,6 1420663 pipe python 25080 root 95w FIFO 0,6 1420663 pipe python 25080 root 96w FIFO 0,6 1428099 pipe python 25080 root 97r FIFO 0,6 1428128 pipe python 25080 root 98w FIFO 0,6 1428128 pipe python 25080 root 99r FIFO 0,6 1433361 pipe python 25080 root 100r FIFO 0,6 1438698 pipe python 25080 root 101r FIFO 0,6 1438742 pipe python 25080 root 102r FIFO 0,6 1438716 pipe python 25080 root 103w FIFO 0,6 1438716 pipe python 25080 root 104w FIFO 0,6 1433361 pipe python 25080 root 105r FIFO 0,6 1438721 pipe python 25080 root 106w FIFO 0,6 1438698 pipe python 25080 root 107w FIFO 0,6 1438721 pipe python 25080 root 108r FIFO 0,6 1438726 pipe python 25080 root 109w FIFO 0,6 1438726 pipe python 25080 root 110r FIFO 0,6 1438731 pipe python 25080 root 111w FIFO 0,6 1438731 pipe python 25080 root 112w FIFO 0,6 1438742 pipe python 25080 root 113r FIFO 0,6 1439782 pipe python 25080 root 114r FIFO 0,6 1439770 pipe python 25080 root 115w FIFO 0,6 1439770 pipe python 25080 root 116r FIFO 0,6 1443874 pipe python 25080 root 117r FIFO 0,6 1439790 pipe python 25080 root 118w FIFO 0,6 1439790 pipe python 25080 root 119r FIFO 0,6 1450126 pipe python 25080 root 120w FIFO 0,6 1439782 pipe python 25080 root 121r FIFO 0,6 1450140 pipe python 25080 root 122w FIFO 0,6 1450140 pipe python 25080 root 123r FIFO 0,6 1450145 pipe python 25080 root 124w FIFO 0,6 1443874 pipe python 25080 root 125w FIFO 0,6 1450145 pipe python 25080 root 126w FIFO 0,6 1450126 pipe python 25080 root 127r FIFO 0,6 1450150 pipe python 25080 root 128w FIFO 0,6 1450150 pipe python 25080 root 129r FIFO 0,6 1455250 pipe python 25080 root 130r FIFO 0,6 1468132 pipe python 25080 root 131r FIFO 0,6 1468146 pipe python 25080 root 132w FIFO 0,6 1468146 pipe python 25080 root 133r FIFO 0,6 1468151 pipe python 25080 root 134w FIFO 0,6 1455250 pipe python 25080 root 135w FIFO 0,6 1468151 pipe python 25080 root 136w FIFO 0,6 1468132 pipe python 25080 root 137r FIFO 0,6 1468156 pipe python 25080 root 138w FIFO 0,6 1468156 pipe python 25080 root 139r FIFO 0,6 1473274 pipe python 25080 root 140r FIFO 0,6 1479572 pipe python 25080 root 141r FIFO 0,6 1479490 pipe python 25080 root 142w FIFO 0,6 1479490 pipe python 25080 root 143r FIFO 0,6 1514284 pipe python 25080 root 144w FIFO 0,6 1473274 pipe python 25080 root 145r FIFO 0,6 1514265 pipe python 25080 root 146r FIFO 0,6 1477378 pipe python 25080 root 147w FIFO 0,6 1477378 pipe python 25080 root 148w FIFO 0,6 1514265 pipe python 25080 root 149w FIFO 0,6 1514284 pipe python 25080 root 150r FIFO 0,6 1514289 pipe python 25080 root 151w FIFO 0,6 1514289 pipe python 25080 root 152r FIFO 0,6 1514294 pipe python 25080 root 153w FIFO 0,6 1514294 pipe python 25080 root 154r FIFO 0,6 1517354 pipe python 25080 root 155w FIFO 0,6 1517354 pipe python 25080 root 156r FIFO 0,6 1519400 pipe python 25080 root 157r FIFO 0,6 1524745 pipe python 25080 root 158r FIFO 0,6 1524715 pipe python 25080 root 159w FIFO 0,6 1524715 pipe python 25080 root 160r FIFO 0,6 1524734 pipe python 25080 root 161w FIFO 0,6 1519400 pipe python 25080 root 162w FIFO 0,6 1479572 pipe python 25080 root 163r FIFO 0,6 1479577 pipe python 25080 root 164w FIFO 0,6 1479577 pipe python 25080 root 165r FIFO 0,6 1479558 pipe python 25080 root 166w FIFO 0,6 1479558 pipe python 25080 root 167r FIFO 0,6 1479582 pipe python 25080 root 168w FIFO 0,6 1479582 pipe python 25080 root 169r FIFO 0,6 1484686 pipe python 25080 root 170r FIFO 0,6 1490836 pipe python 25080 root 171r FIFO 0,6 1490811 pipe python 25080 root 172w FIFO 0,6 1490811 pipe python 25080 root 173r FIFO 0,6 1490830 pipe python 25080 root 174w FIFO 0,6 1484686 pipe python 25080 root 175w FIFO 0,6 1490830 pipe python 25080 root 176w FIFO 0,6 1490836 pipe python 25080 root 177r FIFO 0,6 1490841 pipe python 25080 root 178w FIFO 0,6 1490841 pipe python 25080 root 179r FIFO 0,6 1495941 pipe python 25080 root 180r FIFO 0,6 1503010 pipe python 25080 root 181r FIFO 0,6 1502990 pipe python 25080 root 182w FIFO 0,6 1502990 pipe python 25080 root 183w FIFO 0,6 1503010 pipe python 25080 root 184w FIFO 0,6 1495941 pipe python 25080 root 185r FIFO 0,6 1503015 pipe python 25080 root 186w FIFO 0,6 1503015 pipe python 25080 root 187r FIFO 0,6 1503020 pipe python 25080 root 188w FIFO 0,6 1503020 pipe python 25080 root 189w FIFO 0,6 1524734 pipe python 25080 root 190r FIFO 0,6 1508125 pipe python 25080 root 191w FIFO 0,6 1508125 pipe python 25080 root 192r FIFO 0,6 1524739 pipe python 25080 root 193w FIFO 0,6 1524739 pipe python 25080 root 194w FIFO 0,6 1524745 pipe python 25080 root 195r FIFO 0,6 1525773 pipe python 25080 root 196r FIFO 0,6 1525872 pipe python 25080 root 197u FIFO 0,6 1537150 pipe python 25080 root 198r FIFO 0,6 1525787 pipe python 25080 root 199w FIFO 0,6 1525787 pipe python 25080 root 200w FIFO 0,6 1525773 pipe python 25080 root 201u FIFO 0,6 1537150 pipe python 25080 root 202u FIFO 0,6 1537157 pipe python 25080 root 203u FIFO 0,6 1537157 pipe python 25080 root 204r FIFO 0,6 1525807 pipe python 25080 root 205w FIFO 0,6 1525807 pipe python 25080 root 206u FIFO 0,6 1537167 pipe python 25080 root 207u FIFO 0,6 1537167 pipe python 25080 root 208u FIFO 0,6 1554634 pipe python 25080 root 209u FIFO 0,6 1549511 pipe python 25080 root 210u FIFO 0,6 1549489 pipe python 25080 root 211u FIFO 0,6 1549489 pipe python 25080 root 212u FIFO 0,6 1549511 pipe python 25080 root 213u FIFO 0,6 1543051 pipe python 25080 root 214u FIFO 0,6 1543051 pipe python 25080 root 215u FIFO 0,6 1549516 pipe python 25080 root 216u FIFO 0,6 1549516 pipe python 25080 root 217u FIFO 0,6 1549521 pipe python 25080 root 218w FIFO 0,6 1525872 pipe python 25080 root 219r FIFO 0,6 1530456 pipe python 25080 root 220r FIFO 0,6 1537120 pipe python 25080 root 221w FIFO 0,6 1549521 pipe python 25080 root 222r FIFO 0,6 1557774 pipe python 25080 root 223u FIFO 0,6 1683740 pipe python 25080 root 224w FIFO 0,6 1530456 pipe python 25080 root 225u FIFO 0,6 1683740 pipe python 25080 root 226w FIFO 0,6 1537120 pipe python 25080 root 227u FIFO 0,6 1683745 pipe python 25080 root 228w FIFO 0,6 1554634 pipe python 25080 root 229u FIFO 0,6 1683745 pipe python 25080 root 230r FIFO 0,6 1556689 pipe python 25080 root 231w FIFO 0,6 1556689 pipe python 25080 root 232u FIFO 0,6 1683722 pipe python 25080 root 233u FIFO 0,6 1683722 pipe python 25080 root 234u FIFO 0,6 1683750 pipe python 25080 root 235u FIFO 0,6 1683750 pipe python 25080 root 236u FIFO 0,6 1685963 pipe python 25080 root 237u FIFO 0,6 1685963 pipe python 25080 root 238u FIFO 0,6 1689255 pipe python 25080 root 239u FIFO 0,6 1689255 pipe python 25080 root 240u FIFO 0,6 1689265 pipe python 25080 root 241u FIFO 0,6 1689283 pipe python 25080 root 242u FIFO 0,6 1689283 pipe python 25080 root 243u FIFO 0,6 1692354 pipe python 25080 root 244r FIFO 0,6 1557810 pipe python 25080 root 245w FIFO 0,6 1557810 pipe python 25080 root 246r FIFO 0,6 1557791 pipe python 25080 root 247w FIFO 0,6 1557791 pipe python 25080 root 248w FIFO 0,6 1557774 pipe python 25080 root 249u FIFO 0,6 1692354 pipe python 25080 root 250u FIFO 0,6 1689265 pipe python 25080 root 251u FIFO 0,6 1695495 pipe python 25080 root 252u FIFO 0,6 1699706 pipe python 25080 root 253r FIFO 0,6 1557821 pipe python 25080 root 254w FIFO 0,6 1557821 pipe python 25080 root 255r FIFO 0,6 1558904 pipe python 25080 root 256u FIFO 0,6 1695512 pipe python 25080 root 257u FIFO 0,6 1695512 pipe python 25080 root 258u FIFO 0,6 1695517 pipe python 25080 root 259u FIFO 0,6 1695495 pipe python 25080 root 260u FIFO 0,6 1695517 pipe python 25080 root 261u FIFO 0,6 1695522 pipe python 25080 root 262u FIFO 0,6 1695522 pipe python 25080 root 263u FIFO 0,6 1699706 pipe python 25080 root 264u FIFO 0,6 1700747 pipe python 25080 root 265u FIFO 0,6 1702203 pipe python 25080 root 266u FIFO 0,6 1702203 pipe python 25080 root 267u FIFO 0,6 1708327 pipe python 25080 root 268w FIFO 0,6 1558904 pipe python 25080 root 269r FIFO 0,6 1560162 pipe python 25080 root 270r FIFO 0,6 1560180 pipe python 25080 root 271r FIFO 0,6 1560174 pipe python 25080 root 272w FIFO 0,6 1560174 pipe python 25080 root 273w FIFO 0,6 1560180 pipe python 25080 root 274w FIFO 0,6 1560162 pipe python 25080 root 275r FIFO 0,6 1560185 pipe python 25080 root 276w FIFO 0,6 1560185 pipe python 25080 root 277r FIFO 0,6 1565285 pipe python 25080 root 278r FIFO 0,6 1572323 pipe python 25080 root 279r FIFO 0,6 1572341 pipe python 25080 root 280r FIFO 0,6 1572335 pipe python 25080 root 281w FIFO 0,6 1572335 pipe python 25080 root 282w FIFO 0,6 1565285 pipe python 25080 root 283w FIFO 0,6 1572341 pipe python 25080 root 284w FIFO 0,6 1572323 pipe python 25080 root 285r FIFO 0,6 1572346 pipe python 25080 root 286w FIFO 0,6 1572346 pipe python 25080 root 287r FIFO 0,6 1583586 pipe python 25080 root 288r FIFO 0,6 1583604 pipe python 25080 root 289r FIFO 0,6 1577446 pipe python 25080 root 290w FIFO 0,6 1577446 pipe python 25080 root 291r FIFO 0,6 1583598 pipe python 25080 root 292w FIFO 0,6 1583598 pipe python 25080 root 293w FIFO 0,6 1583604 pipe python 25080 root 294w FIFO 0,6 1583586 pipe python 25080 root 295r FIFO 0,6 1583609 pipe python 25080 root 296w FIFO 0,6 1583609 pipe python 25080 root 297r FIFO 0,6 1588709 pipe python 25080 root 298r FIFO 0,6 1594029 pipe python 25080 root 299r FIFO 0,6 1594042 pipe python 25080 root 300w FIFO 0,6 1594042 pipe python 25080 root 301r FIFO 0,6 1594047 pipe python 25080 root 302w FIFO 0,6 1588709 pipe python 25080 root 303w FIFO 0,6 1594047 pipe python 25080 root 304w FIFO 0,6 1594029 pipe python 25080 root 305r FIFO 0,6 1594052 pipe python 25080 root 306w FIFO 0,6 1594052 pipe python 25080 root 307r FIFO 0,6 1596094 pipe python 25080 root 308w FIFO 0,6 1596094 pipe python 25080 root 309r FIFO 0,6 1596104 pipe python 25080 root 310r FIFO 0,6 1605320 pipe python 25080 root 311r FIFO 0,6 1596121 pipe python 25080 root 312w FIFO 0,6 1596121 pipe python 25080 root 313r FIFO 0,6 1599187 pipe python 25080 root 314w FIFO 0,6 1596104 pipe python 25080 root 315w FIFO 0,6 1599187 pipe python 25080 root 316r FIFO 0,6 1605338 pipe python 25080 root 317w FIFO 0,6 1605338 pipe python 25080 root 318r FIFO 0,6 1605343 pipe python 25080 root 319w FIFO 0,6 1605343 pipe python 25080 root 320w FIFO 0,6 1605320 pipe python 25080 root 321r FIFO 0,6 1605348 pipe python 25080 root 322w FIFO 0,6 1605348 pipe python 25080 root 323r FIFO 0,6 1616575 pipe python 25080 root 324r FIFO 0,6 1616598 pipe python 25080 root 325r FIFO 0,6 1616592 pipe python 25080 root 326w FIFO 0,6 1616592 pipe python 25080 root 327w FIFO 0,6 1616598 pipe python 25080 root 328r FIFO 0,6 1610448 pipe python 25080 root 329w FIFO 0,6 1610448 pipe python 25080 root 330w FIFO 0,6 1616575 pipe python 25080 root 331r FIFO 0,6 1616603 pipe python 25080 root 332w FIFO 0,6 1616603 pipe python 25080 root 333r FIFO 0,6 1621703 pipe python 25080 root 334r FIFO 0,6 1627042 pipe python 25080 root 335r FIFO 0,6 1627082 pipe python 25080 root 336r FIFO 0,6 1627073 pipe python 25080 root 337w FIFO 0,6 1627073 pipe python 25080 root 338w FIFO 0,6 1621703 pipe python 25080 root 339w FIFO 0,6 1627082 pipe python 25080 root 340w FIFO 0,6 1627042 pipe python 25080 root 341r FIFO 0,6 1627090 pipe python 25080 root 342w FIFO 0,6 1627090 pipe python 25080 root 343r FIFO 0,6 1636845 pipe python 25080 root 344w FIFO 0,6 1636845 pipe python 25080 root 345r FIFO 0,6 1638097 pipe python 25080 root 346r FIFO 0,6 1638116 pipe python 25080 root 347r FIFO 0,6 1632760 pipe python 25080 root 348w FIFO 0,6 1632760 pipe python 25080 root 349w FIFO 0,6 1638116 pipe python 25080 root 350r FIFO 0,6 1638121 pipe python 25080 root 351w FIFO 0,6 1638121 pipe python 25080 root 352w FIFO 0,6 1638097 pipe python 25080 root 353r FIFO 0,6 1638126 pipe python 25080 root 354w FIFO 0,6 1638126 pipe python 25080 root 355r FIFO 0,6 1639150 pipe python 25080 root 356w FIFO 0,6 1639150 pipe python 25080 root 357r FIFO 0,6 1639156 pipe python 25080 root 358w FIFO 0,6 1639156 pipe python 25080 root 359r FIFO 0,6 1643247 pipe python 25080 root 360r FIFO 0,6 1639160 pipe python 25080 root 361w FIFO 0,6 1639160 pipe python 25080 root 362r FIFO 0,6 1644293 pipe python 25080 root 363r FIFO 0,6 1644281 pipe python 25080 root 364w FIFO 0,6 1644281 pipe python 25080 root 365r FIFO 0,6 1644287 pipe python 25080 root 366w FIFO 0,6 1643247 pipe python 25080 root 367w FIFO 0,6 1644287 pipe python 25080 root 368w FIFO 0,6 1644293 pipe python 25080 root 369r FIFO 0,6 1644302 pipe python 25080 root 370w FIFO 0,6 1644302 pipe python 25080 root 371r FIFO 0,6 1649401 pipe python 25080 root 372r FIFO 0,6 1649425 pipe python 25080 root 373r FIFO 0,6 1649419 pipe python 25080 root 374w FIFO 0,6 1649419 pipe python 25080 root 375w FIFO 0,6 1649425 pipe python 25080 root 376w FIFO 0,6 1649401 pipe python 25080 root 377r FIFO 0,6 1649430 pipe python 25080 root 378w FIFO 0,6 1649430 pipe python 25080 root 379r FIFO 0,6 1655573 pipe python 25080 root 380w FIFO 0,6 1655573 pipe python 25080 root 381r FIFO 0,6 1659874 pipe python 25080 root 382r FIFO 0,6 1659892 pipe python 25080 root 383r FIFO 0,6 1654540 pipe python 25080 root 384w FIFO 0,6 1654540 pipe python 25080 root 385w FIFO 0,6 1659892 pipe python 25080 root 386r FIFO 0,6 1659897 pipe python 25080 root 387w FIFO 0,6 1659897 pipe python 25080 root 388w FIFO 0,6 1659874 pipe python 25080 root 389r FIFO 0,6 1659902 pipe python 25080 root 390w FIFO 0,6 1659902 pipe python 25080 root 391r FIFO 0,6 1665240 pipe python 25080 root 392r FIFO 0,6 1669337 pipe python 25080 root 393w FIFO 0,6 1669337 pipe python 25080 root 394r FIFO 0,6 1669454 pipe python 25080 root 395w FIFO 0,6 1669454 pipe python 25080 root 396w FIFO 0,6 1665240 pipe python 25080 root 397r FIFO 0,6 1671711 pipe python 25080 root 398w FIFO 0,6 1671711 pipe python 25080 root 399r FIFO 0,6 1671897 pipe python 25080 root 400w FIFO 0,6 1671897 pipe python 25080 root 401u FIFO 0,6 1708355 pipe python 25080 root 402r FIFO 0,6 1671686 pipe python 25080 root 403w FIFO 0,6 1671686 pipe python 25080 root 404r FIFO 0,6 1677418 pipe python 25080 root 405w FIFO 0,6 1677418 pipe python 25080 root 406w FIFO 0,6 1700747 pipe python 25080 root 407u FIFO 0,6 1708355 pipe python 25080 root 408u FIFO 0,6 1708360 pipe python 25080 root 409u FIFO 0,6 1708360 pipe python 25080 root 410w FIFO 0,6 1708327 pipe python 25080 root 411u FIFO 0,6 1708366 pipe python 25080 root 412u FIFO 0,6 1708366 pipe python 25080 root 413r FIFO 0,6 1710416 pipe python 25080 root 414w FIFO 0,6 1710416 pipe python 25080 root 415r FIFO 0,6 1717886 pipe python 25080 root 416w FIFO 0,6 1717886 pipe python 25080 root 417r FIFO 0,6 1713782 pipe python 25080 root 418w FIFO 0,6 1713782 pipe python 25080 root 419r FIFO 0,6 1717896 pipe python 25080 root 420r FIFO 0,6 1710437 pipe python 25080 root 421w FIFO 0,6 1710437 pipe python 25080 root 422r FIFO 0,6 1720262 pipe python 25080 root 423w FIFO 0,6 1720262 pipe python 25080 root 424r FIFO 0,6 1720267 pipe python 25080 root 425w FIFO 0,6 1720267 pipe python 25080 root 426w FIFO 0,6 1717896 pipe python 25080 root 427r FIFO 0,6 1717975 pipe python 25080 root 428w FIFO 0,6 1717975 pipe python 25080 root 429r FIFO 0,6 1720248 pipe python 25080 root 430w FIFO 0,6 1720248 pipe python 25080 root 431r FIFO 0,6 1720279 pipe python 25080 root 432w FIFO 0,6 1720279 pipe python 25080 root 433r FIFO 0,6 1722779 pipe python 25080 root 434w FIFO 0,6 1722779 pipe python 25080 root 435r FIFO 0,6 1722797 pipe python 25080 root 436w FIFO 0,6 1722797 pipe python 25080 root 437r FIFO 0,6 1724010 pipe python 25080 root 438w FIFO 0,6 1724010 pipe python 25080 root 439r FIFO 0,6 1724015 pipe python 25080 root 440w FIFO 0,6 1724015 pipe python 25080 root 441r FIFO 0,6 1725039 pipe python 25080 root 442w FIFO 0,6 1725039 pipe python 25080 root 443r FIFO 0,6 1726070 pipe python 25080 root 444r FIFO 0,6 1731052 pipe python 25080 root 445w FIFO 0,6 1731052 pipe python 25080 root 446r FIFO 0,6 1733138 pipe python 25080 root 447w FIFO 0,6 1733138 pipe python 25080 root 448w FIFO 0,6 1726070 pipe python 25080 root 449r FIFO 0,6 1733143 pipe python 25080 root 450w FIFO 0,6 1733143 pipe python 25080 root 451r FIFO 0,6 1733119 pipe python 25080 root 452w FIFO 0,6 1733119 pipe python 25080 root 453r FIFO 0,6 1733216 pipe python 25080 root 454r FIFO 0,6 1733148 pipe python 25080 root 455w FIFO 0,6 1733148 pipe python 25080 root 456r FIFO 0,6 1734250 pipe python 25080 root 457w FIFO 0,6 1734250 pipe python 25080 root 458r FIFO 0,6 1738338 pipe python 25080 root 459r FIFO 0,6 1742429 pipe python 25080 root 460r FIFO 0,6 1744479 pipe python 25080 root 461w FIFO 0,6 1744479 pipe python 25080 root 462r FIFO 0,6 1744485 pipe python 25080 root 463w FIFO 0,6 1738338 pipe python 25080 root 464w FIFO 0,6 1744485 pipe python 25080 root 465w FIFO 0,6 1742429 pipe python 25080 root 466r FIFO 0,6 1744496 pipe python 25080 root 467u IPv4 1761150 UDP *:1100 python 25080 root 468u unix 0xf50388c0 1761151 socket python 25080 root 469u unix 0xf5039880 1761152 socket python 25080 root 470w FIFO 0,6 1761153 pipe python 25080 root 471w FIFO 0,6 1744496 pipe python 25080 root 472w FIFO 0,6 1733216 pipe python 25080 root 473r FIFO 0,6 1761167 pipe python 25080 root 474r FIFO 0,6 1745591 pipe python 25080 root 475w FIFO 0,6 1745591 pipe python 25080 root 476r FIFO 0,6 1761181 pipe python 25080 root 477w FIFO 0,6 1761181 pipe python 25080 root 478r FIFO 0,6 1761186 pipe python 25080 root 479w FIFO 0,6 1761186 pipe python 25080 root 480w FIFO 0,6 1761167 pipe python 25080 root 481r FIFO 0,6 1762234 pipe python 25080 root 482r FIFO 0,6 1762217 pipe python 25080 root 483w FIFO 0,6 1762217 pipe python 25080 root 484w FIFO 0,6 1762234 pipe python 25080 root 485r FIFO 0,6 1765549 pipe python 25080 root 486w FIFO 0,6 1765549 pipe python 25080 root 487r FIFO 0,6 1767859 pipe python 25080 root 488w FIFO 0,6 1767859 pipe python 25080 root 489r FIFO 0,6 1762247 pipe python 25080 root 490w FIFO 0,6 1762247 pipe python 25080 root 491r FIFO 0,6 1762252 pipe python 25080 root 492w FIFO 0,6 1762252 pipe python 25080 root 493r FIFO 0,6 1767868 pipe python 25080 root 494w FIFO 0,6 1767868 pipe python 25080 root 495r FIFO 0,6 1765559 pipe python 25080 root 496w FIFO 0,6 1765559 pipe python 25080 root 497r FIFO 0,6 1767877 pipe python 25080 root 498w FIFO 0,6 1767877 pipe python 25080 root 499r FIFO 0,6 1767890 pipe python 25080 root 500w FIFO 0,6 1767890 pipe python 25080 root 501r FIFO 0,6 1768917 pipe python 25080 root 502w FIFO 0,6 1768917 pipe python 25080 root 503r FIFO 0,6 1768926 pipe python 25080 root 504w FIFO 0,6 1768926 pipe python 25080 root 505r FIFO 0,6 1768939 pipe python 25080 root 506r FIFO 0,6 1768976 pipe python 25080 root 507w FIFO 0,6 1768976 pipe python 25080 root 508r FIFO 0,6 1768961 pipe python 25080 root 509w FIFO 0,6 1768961 pipe python 25080 root 510w FIFO 0,6 1768939 pipe python 25080 root 511r FIFO 0,6 1768953 pipe python 25080 root 512w FIFO 0,6 1768953 pipe python 25080 root 513r FIFO 0,6 1770003 pipe python 25080 root 514w FIFO 0,6 1770003 pipe python 25080 root 515r FIFO 0,6 1770012 pipe python 25080 root 516w FIFO 0,6 1770012 pipe python 25080 root 517r FIFO 0,6 1770021 pipe python 25080 root 518w FIFO 0,6 1770021 pipe python 25080 root 519r FIFO 0,6 1770026 pipe python 25080 root 520w FIFO 0,6 1770026 pipe python 25080 root 521r FIFO 0,6 1770031 pipe python 25080 root 522w FIFO 0,6 1770031 pipe python 25080 root 523r FIFO 0,6 1771058 pipe python 25080 root 524w FIFO 0,6 1771058 pipe python 25080 root 525r FIFO 0,6 1771074 pipe python 25080 root 526w FIFO 0,6 1771074 pipe python 25080 root 527r FIFO 0,6 1771178 pipe python 25080 root 528w FIFO 0,6 1771178 pipe python 25080 root 529r FIFO 0,6 1771187 pipe python 25080 root 530w FIFO 0,6 1771187 pipe python 25080 root 531r FIFO 0,6 1772214 pipe python 25080 root 532w FIFO 0,6 1772214 pipe python 25080 root 533r FIFO 0,6 1776295 pipe python 25080 root 534w FIFO 0,6 1776295 pipe python 25080 root 535r FIFO 0,6 1776301 pipe python 25080 root 536w FIFO 0,6 1776301 pipe python 25080 root 537r FIFO 0,6 1786584 pipe python 25080 root 538r FIFO 0,6 1786577 pipe python 25080 root 539w FIFO 0,6 1786577 pipe python 25080 root 540w FIFO 0,6 1786584 pipe python 25080 root 541r FIFO 0,6 1776311 pipe python 25080 root 542w FIFO 0,6 1776311 pipe python 25080 root 543r FIFO 0,6 1789726 pipe python 25080 root 544r FIFO 0,6 1805259 pipe python 25080 root 545w FIFO 0,6 1805259 pipe python 25080 root 546r FIFO 0,6 1807301 pipe python 25080 root 547w FIFO 0,6 1807301 pipe python 25080 root 548r FIFO 0,6 1787613 pipe python 25080 root 549w FIFO 0,6 1787613 pipe python 25080 root 550w FIFO 0,6 1789726 pipe python 25080 root 551r FIFO 0,6 1791841 pipe python 25080 root 552w FIFO 0,6 1791841 pipe python 25080 root 553r FIFO 0,6 1796035 pipe python 25080 root 554w FIFO 0,6 1796035 pipe python 25080 root 555r FIFO 0,6 1797139 pipe python 25080 root 556w FIFO 0,6 1797139 pipe python 25080 root 557r FIFO 0,6 1813506 pipe python 25080 root 558w FIFO 0,6 1813506 pipe python 25080 root 559r FIFO 0,6 1815553 pipe python 25080 root 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pipe python 25080 root 760r FIFO 0,6 1947748 pipe python 25080 root 761w FIFO 0,6 1947748 pipe python 25080 root 762r FIFO 0,6 1947775 pipe python 25080 root 763w FIFO 0,6 1947775 pipe python 25080 root 764r FIFO 0,6 1953911 pipe python 25080 root 765w FIFO 0,6 1953911 pipe python 25080 root 766r FIFO 0,6 1959014 pipe python 25080 root 767r FIFO 0,6 1959032 pipe python 25080 root 768w FIFO 0,6 1953893 pipe python 25080 root 769w FIFO 0,6 1959032 pipe python 25080 root 770r FIFO 0,6 1959037 pipe python 25080 root 771w FIFO 0,6 1959037 pipe python 25080 root 772w FIFO 0,6 1959014 pipe python 25080 root 773r FIFO 0,6 1959042 pipe python 25080 root 774w FIFO 0,6 1959042 pipe python 25080 root 775r FIFO 0,6 1960068 pipe python 25080 root 776w FIFO 0,6 1960068 pipe python 25080 root 777r FIFO 0,6 1961092 pipe python 25080 root 778w FIFO 0,6 1961092 pipe python 25080 root 779r FIFO 0,6 1962119 pipe python 25080 root 780w FIFO 0,6 1962119 pipe python 25080 root 781u FIFO 0,6 1970744 pipe python 25080 root 782r FIFO 0,6 1970478 pipe python 25080 root 783w FIFO 0,6 1970478 pipe python 25080 root 784u FIFO 0,6 1970744 pipe python 25080 root 785u FIFO 0,6 1970749 pipe python 25080 root 786r FIFO 0,6 1965183 pipe python 25080 root 787w FIFO 0,6 1965183 pipe python 25080 root 788u FIFO 0,6 1970749 pipe python 25080 root 789u FIFO 0,6 1970754 pipe python 25080 root 790w FIFO 0,6 1970754 pipe python 25080 root 791r FIFO 0,6 1988530 pipe python 25080 root 792r FIFO 0,6 1988548 pipe python 25080 root 793w FIFO 0,6 1988548 pipe python 25080 root 794r FIFO 0,6 1988553 pipe python 25080 root 795w FIFO 0,6 1988553 pipe python 25080 root 796r FIFO 0,6 1976412 pipe python 25080 root 797w FIFO 0,6 1976412 pipe python 25080 root 798w FIFO 0,6 1988530 pipe python 25080 root 799r FIFO 0,6 1988558 pipe python 25080 root 800w FIFO 0,6 1988558 pipe From sitanshu at gmail.com Tue Sep 22 21:46:05 2009 From: sitanshu at gmail.com (Sitanshu Nanavati) Date: Wed, 23 Sep 2009 01:16:05 +0530 Subject: [gst-devel] Uglyplugin building Message-ID: Hello, I am trying to build ugly plugins on gstreamer-0.10.23. I get following errors while doing make. Could anyone please let me know what I am missing? TIA. -s [root at localhost gst-plugins-ugly-0.10.12]# make Making all in gst-libs Making all in gst Making all in gst Making all in asfdemux CC gstasf.o In file included from /usr/include/gstreamer-0.10/gst/rtp/gstbasertpdepayload.h: 24, from gstrtpasfdepay.h:27, from gstasf.c:30: /usr/include/gstreamer-0.10/gst/rtp/gstrtpbuffer.h:54: error: expected ')' befor e '*' token /usr/include/gstreamer-0.10/gst/rtp/gstrtpbuffer.h:75: error: expected ')' befor e '*' token /usr/include/gstreamer-0.10/gst/rtp/gstrtpbuffer.h:77: error: expected ')' befor e '*' token /usr/include/gstreamer-0.10/gst/rtp/gstrtpbuffer.h:87: error: expected ')' befor e '*' token /usr/include/gstreamer-0.10/gst/rtp/gstrtpbuffer.h:89: error: expected ')' befor e '*' token /usr/include/gstreamer-0.10/gst/rtp/gstrtpbuffer.h:92: error: expected ')' befor e '*' token /usr/include/gstreamer-0.10/gst/rtp/gstrtpbuffer.h:94: error: expected ')' befor e '*' token /usr/include/gstreamer-0.10/gst/rtp/gstrtpbuffer.h:97: error: expected ')' befor e '*' token /usr/include/gstreamer-0.10/gst/rtp/gstrtpbuffer.h:99: error: expected ')' befor e '*' token /usr/include/gstreamer-0.10/gst/rtp/gstrtpbuffer.h:105: error: expected ')' befo re '*' token make[3]: *** [libgstasf_la-gstasf.lo] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 [root at localhost gst-plugins-ugly-0.10.12] -------------- next part -------------- An HTML attachment was scrubbed... URL: From t.i.m at zen.co.uk Wed Sep 23 11:20:44 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Wed, 23 Sep 2009 10:20:44 +0100 Subject: [gst-devel] Uglyplugin building In-Reply-To: References: Message-ID: <1253697644.4834.1.camel@zingle> On Wed, 2009-09-23 at 01:16 +0530, Sitanshu Nanavati wrote: Hi, > I am trying to build ugly plugins on gstreamer-0.10.23. I get > following errors while doing make. Could anyone please let me know > what I am missing? > > Making all in asfdemux > CC gstasf.o > In file included > from /usr/include/gstreamer-0.10/gst/rtp/gstbasertpdepayload.h: > 24, > from gstrtpasfdepay.h:27, > from gstasf.c:30: > /usr/include/gstreamer-0.10/gst/rtp/gstrtpbuffer.h:54: error: expected > ')' befor > e '*' token > /usr/include/gstreamer-0.10/gst/rtp/gstrtpbuffer.h:75: error: expected > ')' befor > e '*' token that looks like you are building against gst-plugins-base 0.10.24, which requires gstreamer core 0.10.24 or later. Cheers -Tim From kpawan at gmail.com Wed Sep 23 17:12:29 2009 From: kpawan at gmail.com (Kumar, Pawan) Date: Wed, 23 Sep 2009 20:42:29 +0530 Subject: [gst-devel] Gstreamer for Power PC !! Message-ID: Hi All, Can someone let me know how to compile Gstreamer for PPC ? Regards, /Pawan -------------- next part -------------- An HTML attachment was scrubbed... URL: From yorgasor at gmail.com Wed Sep 23 20:10:00 2009 From: yorgasor at gmail.com (Ron Yorgason) Date: Wed, 23 Sep 2009 14:10:00 -0400 Subject: [gst-devel] sending raw audio Message-ID: <93d1fdd10909231110j181a7bc0w8a2736e8bc6dcdfb@mail.gmail.com> I'm working on an audio/video streaming application on the ARM platform. We don't seem to have enough CPU horsepower on these boards to capture and encode to MP3, so right now we're trying to send raw audio. To capture & stream, I'm using this command: gst-launch-0.10 -v alsasrc ! audio/x-raw-int,rate=24000,width=16,depth=16,channels=1,signed=true ! audioconvert ! rtpL16pay ! udpsink host=192.168.17.81 port=5435 On the receiving side, I have this: gst-launch-0.10 -v udpsrc port=5435 caps ="application/x-rtp, media=(string)audio, clock-rate=(int)24000, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, channel-positions=(int)1, payload=(int)96, ssrc=(guint)1168011267, clock-base=(guint)309599748, seqnum-base=(guint)27324" ! rtpL16depay ! audioconvert ! alsasink sync=false When I test this between my desktop & laptop, it works great. When I go from the ARM board to my desktop, it works ok. But when I go from the ARM board to another ARM board, or my laptop to the ARM board, I hear the speakers turn on, but no sound comes out. The output shows this: Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstSystemClock /GstPipeline:pipeline0/GstRtpL16Depay:rtpl16depay0.GstPad:src: caps = audio/x-raw-int, endianness=(int)4321, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)24000, channels=(int)1, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_NONE > /GstPipeline:pipeline0/GstRtpL16Depay:rtpl16depay0.GstPad:sink: caps = application/x-rtp, media=(string)audio, clock-rate=(int)24000, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, channel-positions=(int)1, payload=(int)96, ssrc=(guint)1168011267, clock-base=(guint)309599748, seqnum-base=(guint)27324 /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)24000, channels=(int)1, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_NONE > /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = audio/x-raw-int, endianness=(int)4321, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)24000, channels=(int)1, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_NONE > /GstPipeline:pipeline0/GstAlsaSink:alsasink0.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)24000, channels=(int)1, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_NONE > I couldn't find a good definition of what the GST_AUDIO_CHANNEL_POSITION_NONE means, and I wasn't able to set it to GST_AUDIO_CHANNEL_POSITION_MONO from the command line (it looks like I need to use python or C APIs to do that), but from what I can tell, this comes from multichannel support, and I have just specified a single audio channel. Is this what is preventing me from hearing the audio? If I capture to a WAV file, and then play it back afterwards, it sounds fine. So I'm not sure why streaming is failing so badly. The playback process also dies with a "Terminated" message within a minute or two. The ARM boards are running gstreamer-0.10.22. I see that there's been a couple revisions since then, and if I have to upgrade to make it work, I will. But I'd rather see if there's a way I can make this version work. --Ron From katcipis at inf.ufsc.br Wed Sep 23 21:22:00 2009 From: katcipis at inf.ufsc.br (Tiago Katcipis) Date: Wed, 23 Sep 2009 16:22:00 -0300 Subject: [gst-devel] Noise remover Message-ID: <60a9403b0909231222u17d80973x7614e611bf08cd2c@mail.gmail.com> Is there any plugin that does noise removal on an audio stream? i searched on gstreamer site and at gst-inspect but the only noise remover that i have found is for v?deo. best regards, Katcipis -------------- next part -------------- An HTML attachment was scrubbed... URL: From sitanshu at gmail.com Wed Sep 23 21:55:39 2009 From: sitanshu at gmail.com (Sitanshu Nanavati) Date: Thu, 24 Sep 2009 01:25:39 +0530 Subject: [gst-devel] Noise remover In-Reply-To: <60a9403b0909231222u17d80973x7614e611bf08cd2c@mail.gmail.com> References: <60a9403b0909231222u17d80973x7614e611bf08cd2c@mail.gmail.com> Message-ID: Hello All, I am trying to run audio thru pipeline. The control stalls at PREROLLING. I see a warning and an error prior to prerolling state. I am not sure from where they are popping and why. Can someone please let me know 1) what is the root cause of below warning and error. Would it harm the audio playback? 2) what i should be checking for the pipeline stalling at PREROLL. Thanks in advance. -s ..... ...... (gst-launch-0.10:19809): GLib-GObject-WARNING **: specified class size for type `MyAudio' is smaller than the parent type's `GObject' class size (gst-launch-0.10:19809): GLib-CRITICAL **: g_once_init_leave: assertion `initial ization_value != 0' failed Pipeline is PREROLLING ... -------------- next part -------------- An HTML attachment was scrubbed... URL: From sitanshu at gmail.com Wed Sep 23 21:57:42 2009 From: sitanshu at gmail.com (Sitanshu Nanavati) Date: Thu, 24 Sep 2009 01:27:42 +0530 Subject: [gst-devel] Pipeline stalls Message-ID: Updated the subject line. Please ignore the earlier message. Hello All, > > I am trying to run audio thru pipeline. The control stalls at PREROLLING. > I see a warning and an error prior to prerolling state. I am not sure from > where they are popping and why. > > Can someone please let me know > 1) what is the root cause of below warning and error. Would it harm the > audio playback? > 2) what i should be checking for the pipeline stalling at PREROLL. > > Thanks in advance. > -s > > ..... > ...... > (gst-launch-0.10:19809): GLib-GObject-WARNING **: specified class size for > type > `MyAudio' is smaller than the parent type's `GObject' class size > > (gst-launch-0.10:19809): GLib-CRITICAL **: g_once_init_leave: assertion > `initial > ization_value != 0' failed > Pipeline is PREROLLING ... > -------------- next part -------------- An HTML attachment was scrubbed... URL: From katcipis at inf.ufsc.br Wed Sep 23 22:13:04 2009 From: katcipis at inf.ufsc.br (Tiago Katcipis) Date: Wed, 23 Sep 2009 17:13:04 -0300 Subject: [gst-devel] Pipeline stalls In-Reply-To: References: Message-ID: <60a9403b0909231313v36599646vcd387de16e24fbd4@mail.gmail.com> can you specify better the pipeline? On Wed, Sep 23, 2009 at 4:57 PM, Sitanshu Nanavati wrote: > Updated the subject line. > Please ignore the earlier message. > > Hello All, >> >> I am trying to run audio thru pipeline. The control stalls at PREROLLING. >> I see a warning and an error prior to prerolling state. I am not sure >> from where they are popping and why. >> >> Can someone please let me know >> 1) what is the root cause of below warning and error. Would it harm the >> audio playback? >> 2) what i should be checking for the pipeline stalling at PREROLL. >> >> Thanks in advance. >> -s >> >> ..... >> ...... >> (gst-launch-0.10:19809): GLib-GObject-WARNING **: specified class size for >> type >> `MyAudio' is smaller than the parent type's `GObject' class size >> >> (gst-launch-0.10:19809): GLib-CRITICAL **: g_once_init_leave: assertion >> `initial >> ization_value != 0' failed >> Pipeline is PREROLLING ... >> > > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em menos de 5 minutos aparecer? algu?m pra discordar de sua escolha de distribui??o, do particionamento, do gerenciador de janelas, do ambiente de desktop, do editor de textos..." -------------- next part -------------- An HTML attachment was scrubbed... URL: From sitanshu at gmail.com Wed Sep 23 22:18:22 2009 From: sitanshu at gmail.com (Sitanshu Nanavati) Date: Thu, 24 Sep 2009 01:48:22 +0530 Subject: [gst-devel] Pipeline stalls In-Reply-To: <60a9403b0909231313v36599646vcd387de16e24fbd4@mail.gmail.com> References: <60a9403b0909231313v36599646vcd387de16e24fbd4@mail.gmail.com> Message-ID: Are you looking for the pipeline construction command-line? $ gst-launch filesrc location=/my/home/audio.wma ! ffdemux_asf ! my_wma_decoder ! my_sink On Thu, Sep 24, 2009 at 1:43 AM, Tiago Katcipis wrote: > can you specify better the pipeline? > > On Wed, Sep 23, 2009 at 4:57 PM, Sitanshu Nanavati wrote: > >> Updated the subject line. >> Please ignore the earlier message. >> >> Hello All, >>> >>> I am trying to run audio thru pipeline. The control stalls at >>> PREROLLING. >>> I see a warning and an error prior to prerolling state. I am not sure >>> from where they are popping and why. >>> >>> Can someone please let me know >>> 1) what is the root cause of below warning and error. Would it harm the >>> audio playback? >>> 2) what i should be checking for the pipeline stalling at PREROLL. >>> >>> Thanks in advance. >>> -s >>> >>> ..... >>> ...... >>> (gst-launch-0.10:19809): GLib-GObject-WARNING **: specified class size >>> for type >>> `MyAudio' is smaller than the parent type's `GObject' class size >>> >>> (gst-launch-0.10:19809): GLib-CRITICAL **: g_once_init_leave: assertion >>> `initial >>> ization_value != 0' failed >>> Pipeline is PREROLLING ... >>> >> >> >> >> ------------------------------------------------------------------------------ >> Come build with us! The BlackBerry® Developer Conference in SF, CA >> is the only developer event you need to attend this year. Jumpstart your >> developing skills, take BlackBerry mobile applications to market and stay >> ahead of the curve. Join us from November 9-12, 2009. Register >> now! >> http://p.sf.net/sfu/devconf >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > -- > "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta > sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em menos > de 5 minutos aparecer? algu?m pra discordar de sua escolha de distribui??o, > do particionamento, do gerenciador de janelas, do ambiente de desktop, do > editor de textos..." > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From msmith at xiph.org Wed Sep 23 22:29:26 2009 From: msmith at xiph.org (Michael Smith) Date: Wed, 23 Sep 2009 13:29:26 -0700 Subject: [gst-devel] Pipeline stalls In-Reply-To: References: <60a9403b0909231313v36599646vcd387de16e24fbd4@mail.gmail.com> Message-ID: <3c1737210909231329u51b53beqa1ab910a9c09ac1c@mail.gmail.com> On Wed, Sep 23, 2009 at 1:18 PM, Sitanshu Nanavati wrote: > Are you looking for the pipeline construction command-line? > $ gst-launch filesrc location=/my/home/audio.wma ! ffdemux_asf ! > my_wma_decoder ! my_sink You shouldn't use ffdemux_asf. Use asfdemux instead. If it _still_ doesn't work, then it's almost certainly a bug in your wma decoder plugin - that we don't have any information about, so we can't help you with it. The glib assertion failures you showed suggested that you didn't create/register the gobject for your WMA decoder element correctly. Mike From katcipis at inf.ufsc.br Wed Sep 23 22:35:14 2009 From: katcipis at inf.ufsc.br (Tiago Katcipis) Date: Wed, 23 Sep 2009 17:35:14 -0300 Subject: [gst-devel] Pipeline stalls In-Reply-To: References: <60a9403b0909231313v36599646vcd387de16e24fbd4@mail.gmail.com> Message-ID: <60a9403b0909231335r2236bb2ax7a72f033e8d31327@mail.gmail.com> Try to use gst-debug-level=5 to see all debug information, it helped me a lot on such situations. Without knowing better my_wma_decoder and my_sink is hard to help more. http://gstreamer.freedesktop.org/data/doc/gstreamer/head/manual/html/section-checklist-debug.html On Wed, Sep 23, 2009 at 5:18 PM, Sitanshu Nanavati wrote: > Are you looking for the pipeline construction command-line? > $ gst-launch filesrc location=/my/home/audio.wma ! ffdemux_asf ! > my_wma_decoder ! my_sink > > On Thu, Sep 24, 2009 at 1:43 AM, Tiago Katcipis wrote: > >> can you specify better the pipeline? >> >> On Wed, Sep 23, 2009 at 4:57 PM, Sitanshu Nanavati wrote: >> >>> Updated the subject line. >>> Please ignore the earlier message. >>> >>> Hello All, >>>> >>>> I am trying to run audio thru pipeline. The control stalls at >>>> PREROLLING. >>>> I see a warning and an error prior to prerolling state. I am not sure >>>> from where they are popping and why. >>>> >>>> Can someone please let me know >>>> 1) what is the root cause of below warning and error. Would it harm >>>> the audio playback? >>>> 2) what i should be checking for the pipeline stalling at PREROLL. >>>> >>>> Thanks in advance. >>>> -s >>>> >>>> ..... >>>> ...... >>>> (gst-launch-0.10:19809): GLib-GObject-WARNING **: specified class size >>>> for type >>>> `MyAudio' is smaller than the parent type's `GObject' class size >>>> >>>> (gst-launch-0.10:19809): GLib-CRITICAL **: g_once_init_leave: assertion >>>> `initial >>>> ization_value != 0' failed >>>> Pipeline is PREROLLING ... >>>> >>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Come build with us! The BlackBerry® Developer Conference in SF, CA >>> is the only developer event you need to attend this year. Jumpstart your >>> developing skills, take BlackBerry mobile applications to market and stay >>> ahead of the curve. Join us from November 9-12, 2009. Register >>> now! >>> http://p.sf.net/sfu/devconf >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >> >> >> -- >> "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta >> sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em menos >> de 5 minutos aparecer? algu?m pra discordar de sua escolha de distribui??o, >> do particionamento, do gerenciador de janelas, do ambiente de desktop, do >> editor de textos..." >> >> >> ------------------------------------------------------------------------------ >> Come build with us! The BlackBerry® Developer Conference in SF, CA >> is the only developer event you need to attend this year. Jumpstart your >> developing skills, take BlackBerry mobile applications to market and stay >> ahead of the curve. Join us from November 9-12, 2009. Register >> now! >> http://p.sf.net/sfu/devconf >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em menos de 5 minutos aparecer? algu?m pra discordar de sua escolha de distribui??o, do particionamento, do gerenciador de janelas, do ambiente de desktop, do editor de textos..." -------------- next part -------------- An HTML attachment was scrubbed... URL: From ensonic at hora-obscura.de Wed Sep 23 22:40:48 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Wed, 23 Sep 2009 23:40:48 +0300 Subject: [gst-devel] Noise remover In-Reply-To: <60a9403b0909231222u17d80973x7614e611bf08cd2c@mail.gmail.com> References: <60a9403b0909231222u17d80973x7614e611bf08cd2c@mail.gmail.com> Message-ID: <4ABA87D0.6030103@hora-obscura.de> Tiago Katcipis schrieb: > Is there any plugin that does noise removal on an audio stream? i searched > on gstreamer site and at gst-inspect but the only noise remover that i have > found is for v?deo. We don't have one in the gst-plugins- sets. Also the algorithm depends a bit what kind of noise you need to get rid of. There is a programm called GnomeWaveCleaner that has some algorithms that might be good candidates for gstreamer plugins. You could also check if you find a ladspa or lv2 plugin and use that via the bridges. Stefan > > best regards, > Katcipis > > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > > > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From katcipis at inf.ufsc.br Wed Sep 23 22:58:10 2009 From: katcipis at inf.ufsc.br (Tiago Katcipis) Date: Wed, 23 Sep 2009 17:58:10 -0300 Subject: [gst-devel] Noise remover In-Reply-To: <4ABA87D0.6030103@hora-obscura.de> References: <60a9403b0909231222u17d80973x7614e611bf08cd2c@mail.gmail.com> <4ABA87D0.6030103@hora-obscura.de> Message-ID: <60a9403b0909231358i6497d6avc1e91d11a88d17ee@mail.gmail.com> On Wed, Sep 23, 2009 at 5:40 PM, Stefan Kost wrote: > Tiago Katcipis schrieb: > > Is there any plugin that does noise removal on an audio stream? i > searched > > on gstreamer site and at gst-inspect but the only noise remover that i > have > > found is for v?deo. > > We don't have one in the gst-plugins- sets. Also the algorithm depends a > bit > what kind of noise you need to get rid of. There is a programm called > GnomeWaveCleaner that has some algorithms that might be good candidates for > gstreamer plugins. You could also check if you find a ladspa or lv2 plugin > and > use that via the bridges. > > Stefan > Thanks Stefan, ill take a look. > > > > > best regards, > > Katcipis > > > > > > > > ------------------------------------------------------------------------ > > > > > ------------------------------------------------------------------------------ > > Come build with us! The BlackBerry® Developer Conference in SF, CA > > is the only developer event you need to attend this year. Jumpstart your > > developing skills, take BlackBerry mobile applications to market and stay > > ahead of the curve. Join us from November 9-12, 2009. Register > now! > > http://p.sf.net/sfu/devconf > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em menos de 5 minutos aparecer? algu?m pra discordar de sua escolha de distribui??o, do particionamento, do gerenciador de janelas, do ambiente de desktop, do editor de textos..." -------------- next part -------------- An HTML attachment was scrubbed... URL: From fabioestevam at yahoo.com Thu Sep 24 00:41:29 2009 From: fabioestevam at yahoo.com (Fabio Estevam) Date: Wed, 23 Sep 2009 15:41:29 -0700 (PDT) Subject: [gst-devel] Playbin2 to force ximagesink and color conversion Message-ID: <319796.52430.qm@web51008.mail.re2.yahoo.com> Hi, The following pipeline works fine on my system: gst-launch-0.10 -v filesrc location=file.mp4 ! decodebin ! ffmpegcolorspace ! ximagesink On a C application I would like to use playbin2 and force ffmpegcolorspace and ximagesink in the pipeline. Is this possible? If so, does anyone have any example? Thanks, Fabio Estevam From msmith at xiph.org Thu Sep 24 00:45:19 2009 From: msmith at xiph.org (Michael Smith) Date: Wed, 23 Sep 2009 15:45:19 -0700 Subject: [gst-devel] Playbin2 to force ximagesink and color conversion In-Reply-To: <319796.52430.qm@web51008.mail.re2.yahoo.com> References: <319796.52430.qm@web51008.mail.re2.yahoo.com> Message-ID: <3c1737210909231545q529a91a3u239b4df4060136a1@mail.gmail.com> On Wed, Sep 23, 2009 at 3:41 PM, Fabio Estevam wrote: > Hi, > > The following pipeline works fine on my system: > > gst-launch-0.10 -v filesrc location=file.mp4 ! decodebin ! ffmpegcolorspace ! ximagesink > > On a C application I would like to use playbin2 and force ffmpegcolorspace and ximagesink in the pipeline. > > Is this possible? If so, does anyone have any example? playbin2 will always use ffmpegcolorspace, no need to do anything special. If you want to force usage of a particular sink, you can create the sink you want, then set it as the video-sink property on playbin2. Mike From ds at entropywave.com Thu Sep 24 04:36:58 2009 From: ds at entropywave.com (David Schleef) Date: Wed, 23 Sep 2009 19:36:58 -0700 Subject: [gst-devel] Gstreamer for Power PC !! In-Reply-To: References: Message-ID: <20090924023658.GA29302@entropywave.com> On Wed, Sep 23, 2009 at 08:42:29PM +0530, Kumar, Pawan wrote: > Can someone let me know how to compile Gstreamer for PPC ? Just like every other platform: ./configure make make install dave... From marc.leeman at gmail.com Thu Sep 24 10:26:19 2009 From: marc.leeman at gmail.com (Marc Leeman) Date: Thu, 24 Sep 2009 10:26:19 +0200 Subject: [gst-devel] Gstreamer for Power PC !! In-Reply-To: <20090924023658.GA29302@entropywave.com> References: <20090924023658.GA29302@entropywave.com> Message-ID: <20090924082618.GA31885@crichton.homelinux.org> > > Can someone let me know how to compile Gstreamer for PPC ? > > Just like every other platform: > > ./configure > make > make install :-) I wasn't planning on touching this request that seems to do a real effort in omitting critical information. I would assume (but that just might be me) that he's talking about building for PowerPC on a non-PowerPC build environment; and once we've established that; it might even be that he's investigating a different libc than the typical one. I don't even want to consider the possibility of packaging. In that case; this is 'typical crosscompiling' and would probably not a topic for this mailing list. -- greetz, marc We must believe that it is the darkest before the dawn of a beautiful new world. We will see it when we believe it. -- Saul Alinsky crichton 2.6.26 #1 PREEMPT Tue Jul 29 21:17:59 CDT 2008 GNU/Linux -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: Digital signature URL: From rohan at perzonae.com Thu Sep 24 11:15:57 2009 From: rohan at perzonae.com (Rohan) Date: Thu, 24 Sep 2009 11:15:57 +0200 Subject: [gst-devel] creating bins and ghostpads Message-ID: <4ABB38CD.902@perzonae.com> Hi all, I am obviously doing something wrong here, but cannot figure out what. Essentially the pipeline is the same as the receiver but I have broken it into bins. I seem to be having problems with the ghostpad, because I get this error: (stream_receiver.py:1914): GStreamer-WARNING **: Trying to connect elements that don't share a common ancestor: vidsource and vidbin Traceback (most recent call last): File "stream_receiver.py", line 118, in server() File "stream_receiver.py", line 27, in __init__ vidsource.link(vidbin) gst.LinkError: failed to link vidsource with vidbin Here is the code: ---------------------------------------------------------------------- #!/bin/env python # The gst pipeline # pipe="udpsrc port=5000 ! smokedec ! autovideosink" # # audio # pipe="$pipe tcpclientsrc host=127.0.0.1 port=5001 ! " # pipe="$pipe speexdec ! queue ! alsasink sync=false " import sys,os import gobject import pygst pygst.require("0.10") import gst class server (object): def __init__(self): # start with video self.pipe = gst.Pipeline("player") # video source vidsource = gst.element_factory_make("udpsrc", "vidsource") vidsource.set_property("port", 5000) vidbin = self.buildvid() self.pipe.add(vidbin) vidsource.link(vidbin) self.pipe.set_state(gst.STATE_PLAYING) def buildvid(self): bin = gst.Bin("vidbin") queue = gst.element_factory_make("queue") smokedec = gst.element_factory_make("smokedec") vidsink = gst.element_factory_make("autovideosink") bin.add(queue, smokedec, vidsink) gst.element_link_many(queue, smokedec, vidsink) # ghostpad binsink = gst.GhostPad("binsink", queue.get_pad("sink")) bin.add_pad(binsink) return bin if __name__ == "__main__": server() loop = gobject.MainLoop() loop.run() ---------------------------------------------------------------------------- What is driving me slightly nuts is that if I put all the buildvid code into init and have a long series of object instatiations with add to pipeline and link it works, but as soon as I started messing with ghostpads and bin (which makes the code much more manageable when adding functionality for windows and macs) this messed up. Below is more code that works, and I cannot figure out why this and why not that. And the other thing is this does work in a commandline pipeline such as the one sitting at the top of the script in comments. I am using the same system for the sender, but I'll only include the local display. This sends an image happily, using the build_localvid bin, and linking it to the camera. ----------------------------------------------------------------------------- #!/bin/env python import sys, os import gobject import pygst pygst.require("0.10") import gst class client(object): def __init__(self): self.pipe = gst.Pipeline("sender") # Initial video input camera = gst.element_factory_make("v4l2src", "camera") vidtee = gst.element_factory_make("tee", "vidtee") self.pipe.add(camera, vidtee) camera.link(vidtee) # local video localvidbin = self.build_localvid() self.pipe.add(localvidbin) vidtee.link(localvidbin) self.pipe.set_state(gst.STATE_PLAYING) def build_localvid(self): """This bin takes a camera (video) stream and produces a live image locally.""" bin = gst.Bin("localvid") queue = gst.element_factory_make("queue") out = gst.element_factory_make("xvimagesink") bin.add(out, queue) queue.link_pads("src", out, "sink") binsink = gst.GhostPad("binsink", queue.get_pad("sink")) bin.add_pad(binsink) return bin if __name__ == '__main__': client() loop = gobject.MainLoop() loop.run() --------------------------------------------------------------------------- This does the correct thing, and I get a live image from the camera. I am completely stumped on this one, and have resorted to using gst.parse_bin_from_description() with the gst-launch pipeline, but this is far from ideal from a maintenance perspective. :) I am sure I am missing something pretty obvious, but so far flailing around in my ignorance has not let me stumble on a solution. Thanks for any help, Rohan From gstelzz at yahoo.fr Thu Sep 24 11:28:57 2009 From: gstelzz at yahoo.fr (Aurelien Grimaud) Date: Thu, 24 Sep 2009 11:28:57 +0200 Subject: [gst-devel] creating bins and ghostpads In-Reply-To: <4ABB38CD.902@perzonae.com> References: <4ABB38CD.902@perzonae.com> Message-ID: <4ABB3BD9.7090206@yahoo.fr> Add vidsource in pipeline ? Rohan a ?crit : > Hi all, > > I am obviously doing something wrong here, but cannot figure out > what. > > Essentially the pipeline is the same as the receiver but I have broken > it into bins. I seem to be having problems with the ghostpad, because > I get this error: > > (stream_receiver.py:1914): GStreamer-WARNING **: Trying to connect elements that > don't share a common ancestor: vidsource and vidbin > > Traceback (most recent call last): > File "stream_receiver.py", line 118, in > server() > File "stream_receiver.py", line 27, in __init__ > vidsource.link(vidbin) > gst.LinkError: failed to link vidsource with vidbin > > Here is the code: > > ---------------------------------------------------------------------- > #!/bin/env python > > # The gst pipeline > # pipe="udpsrc port=5000 ! smokedec ! autovideosink" > # # audio > # pipe="$pipe tcpclientsrc host=127.0.0.1 port=5001 ! " > # pipe="$pipe speexdec ! queue ! alsasink sync=false " > > import sys,os > import gobject > import pygst > pygst.require("0.10") > import gst > > class server (object): > def __init__(self): > # start with video > > self.pipe = gst.Pipeline("player") > > # video source > vidsource = gst.element_factory_make("udpsrc", "vidsource") > vidsource.set_property("port", 5000) > > vidbin = self.buildvid() > self.pipe.add(vidbin) > vidsource.link(vidbin) > > > self.pipe.set_state(gst.STATE_PLAYING) > > def buildvid(self): > bin = gst.Bin("vidbin") > queue = gst.element_factory_make("queue") > smokedec = gst.element_factory_make("smokedec") > vidsink = gst.element_factory_make("autovideosink") > bin.add(queue, smokedec, vidsink) > gst.element_link_many(queue, smokedec, vidsink) > > # ghostpad > binsink = gst.GhostPad("binsink", queue.get_pad("sink")) > bin.add_pad(binsink) > return bin > > if __name__ == "__main__": > server() > loop = gobject.MainLoop() > loop.run() > > ---------------------------------------------------------------------------- > > What is driving me slightly nuts is that if I put all the buildvid code > into init and have a long series of object instatiations with add to > pipeline and link it works, but as soon as I started messing with > ghostpads and bin (which makes the code much more manageable when > adding functionality for windows and macs) this messed up. Below is > more code that works, and I cannot figure out why this and why not > that. > > And the other thing is this does work in a commandline pipeline such > as the one sitting at the top of the script in comments. > > I am using the same system for the sender, but I'll only include the > local display. This sends an image happily, using the build_localvid > bin, and linking it to the camera. > > ----------------------------------------------------------------------------- > #!/bin/env python > > import sys, os > import gobject > import pygst > pygst.require("0.10") > import gst > > class client(object): > > def __init__(self): > self.pipe = gst.Pipeline("sender") > > # Initial video input > camera = gst.element_factory_make("v4l2src", "camera") > vidtee = gst.element_factory_make("tee", "vidtee") > self.pipe.add(camera, vidtee) > camera.link(vidtee) > > # local video > localvidbin = self.build_localvid() > self.pipe.add(localvidbin) > vidtee.link(localvidbin) > > self.pipe.set_state(gst.STATE_PLAYING) > > def build_localvid(self): > """This bin takes a camera (video) stream and produces a live image > locally.""" > bin = gst.Bin("localvid") > queue = gst.element_factory_make("queue") > out = gst.element_factory_make("xvimagesink") > bin.add(out, queue) > queue.link_pads("src", out, "sink") > binsink = gst.GhostPad("binsink", queue.get_pad("sink")) > bin.add_pad(binsink) > return bin > > if __name__ == '__main__': > client() > loop = gobject.MainLoop() > loop.run() > > --------------------------------------------------------------------------- > > This does the correct thing, and I get a live image from the camera. > > I am completely stumped on this one, and have resorted to using > gst.parse_bin_from_description() with the gst-launch pipeline, but > this is far from ideal from a maintenance perspective. :) > > I am sure I am missing something pretty obvious, but so far flailing > around in my ignorance has not let me stumble on a solution. > > Thanks for any help, > > Rohan > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > From gnusercn at gmail.com Thu Sep 24 11:59:26 2009 From: gnusercn at gmail.com (os user) Date: Thu, 24 Sep 2009 17:59:26 +0800 Subject: [gst-devel] Help about setting GST_LEVEL_DEFAULT at compiling time. Message-ID: <34f262ce0909240259x391ef6eekf2b5aa9a7c35bf84@mail.gmail.com> Hi all, Could someone please tell me what's correct or elegant way to set GST_LEVEL_DEFAULT at compiling time? If I'm not afraid of being annoyed and want to see more debugging information, should I set GST_LEVEL_DEFAULT to GST_LEVEL_COUNT? Thanks. From jam at smru.co.uk Thu Sep 24 12:54:10 2009 From: jam at smru.co.uk (Andy Maginnis) Date: Thu, 24 Sep 2009 11:54:10 +0100 Subject: [gst-devel] sending raw audio In-Reply-To: <93d1fdd10909231110j181a7bc0w8a2736e8bc6dcdfb@mail.gmail.com> References: <93d1fdd10909231110j181a7bc0w8a2736e8bc6dcdfb@mail.gmail.com> Message-ID: Ron, What ARM are you using? We have a cortex A8, inside a OMAP3530 on a Gumstix Overo Water a -----Original Message----- From: Ron Yorgason [mailto:yorgasor at gmail.com] Sent: 23 September 2009 19:10 To: gstreamer-devel at lists.sourceforge.net Subject: [gst-devel] sending raw audio I'm working on an audio/video streaming application on the ARM platform. We don't seem to have enough CPU horsepower on these boards to capture and encode to MP3, so right now we're trying to send raw audio. To capture & stream, I'm using this command: gst-launch-0.10 -v alsasrc ! audio/x-raw-int,rate=24000,width=16,depth=16,channels=1,signed=true ! audioconvert ! rtpL16pay ! udpsink host=192.168.17.81 port=5435 On the receiving side, I have this: gst-launch-0.10 -v udpsrc port=5435 caps ="application/x-rtp, media=(string)audio, clock-rate=(int)24000, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, channel-positions=(int)1, payload=(int)96, ssrc=(guint)1168011267, clock-base=(guint)309599748, seqnum-base=(guint)27324" ! rtpL16depay ! audioconvert ! alsasink sync=false When I test this between my desktop & laptop, it works great. When I go from the ARM board to my desktop, it works ok. But when I go from the ARM board to another ARM board, or my laptop to the ARM board, I hear the speakers turn on, but no sound comes out. The output shows this: Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstSystemClock /GstPipeline:pipeline0/GstRtpL16Depay:rtpl16depay0.GstPad:src: caps = audio/x-raw-int, endianness=(int)4321, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)24000, channels=(int)1, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_NONE > /GstPipeline:pipeline0/GstRtpL16Depay:rtpl16depay0.GstPad:sink: caps = application/x-rtp, media=(string)audio, clock-rate=(int)24000, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, channel-positions=(int)1, payload=(int)96, ssrc=(guint)1168011267, clock-base=(guint)309599748, seqnum-base=(guint)27324 /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)24000, channels=(int)1, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_NONE > /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = audio/x-raw-int, endianness=(int)4321, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)24000, channels=(int)1, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_NONE > /GstPipeline:pipeline0/GstAlsaSink:alsasink0.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)24000, channels=(int)1, channel-positions=(GstAudioChannelPosition)< GST_AUDIO_CHANNEL_POSITION_NONE > I couldn't find a good definition of what the GST_AUDIO_CHANNEL_POSITION_NONE means, and I wasn't able to set it to GST_AUDIO_CHANNEL_POSITION_MONO from the command line (it looks like I need to use python or C APIs to do that), but from what I can tell, this comes from multichannel support, and I have just specified a single audio channel. Is this what is preventing me from hearing the audio? If I capture to a WAV file, and then play it back afterwards, it sounds fine. So I'm not sure why streaming is failing so badly. The playback process also dies with a "Terminated" message within a minute or two. The ARM boards are running gstreamer-0.10.22. I see that there's been a couple revisions since then, and if I have to upgrade to make it work, I will. But I'd rather see if there's a way I can make this version work. --Ron ------------------------------------------------------------------------ ------ Come build with us! The BlackBerry® Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9-12, 2009. Register now! http://p.sf.net/sfu/devconf _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From yorgasor at gmail.com Thu Sep 24 14:26:16 2009 From: yorgasor at gmail.com (Ron Yorgason) Date: Thu, 24 Sep 2009 08:26:16 -0400 Subject: [gst-devel] sending raw audio In-Reply-To: References: <93d1fdd10909231110j181a7bc0w8a2736e8bc6dcdfb@mail.gmail.com> Message-ID: <93d1fdd10909240526u413824b0j90c3074cbe81183d@mail.gmail.com> I have a Freescale iMX27. --Ron On Thu, Sep 24, 2009 at 6:54 AM, Andy Maginnis wrote: > Ron, > What ARM are you using? We have a cortex A8, > inside a OMAP3530 on a Gumstix Overo Water > a > > -----Original Message----- > From: Ron Yorgason [mailto:yorgasor at gmail.com] > Sent: 23 September 2009 19:10 > To: gstreamer-devel at lists.sourceforge.net > Subject: [gst-devel] sending raw audio > > I'm working on an audio/video streaming application on the ARM > platform. ?We don't seem to have enough CPU horsepower on these boards > to capture and encode to MP3, so right now we're trying to send raw > audio. > > To capture & stream, I'm using this command: > > gst-launch-0.10 -v ?alsasrc ! > audio/x-raw-int,rate=24000,width=16,depth=16,channels=1,signed=true ! > audioconvert ! rtpL16pay ?! udpsink host=192.168.17.81 port=5435 > > On the receiving side, I have this: > gst-launch-0.10 -v udpsrc port=5435 caps ="application/x-rtp, > media=(string)audio, clock-rate=(int)24000, encoding-name=(string)L16, > encoding-params=(string)1, channels=(int)1, channel-positions=(int)1, > payload=(int)96, ssrc=(guint)1168011267, clock-base=(guint)309599748, > seqnum-base=(guint)27324" ?! ? rtpL16depay ! ?audioconvert ! alsasink > sync=false > > When I test this between my desktop & laptop, it works great. ?When I > go from the ARM board to my desktop, it works ok. ?But when I go from > the ARM board to another ARM board, or my laptop to the ARM board, I > hear the speakers turn on, but no sound comes out. ?The output shows > this: > > Setting pipeline to PAUSED ... > Pipeline is live and does not need PREROLL ... > Setting pipeline to PLAYING ... > New clock: GstSystemClock > /GstPipeline:pipeline0/GstRtpL16Depay:rtpl16depay0.GstPad:src: caps = > audio/x-raw-int, endianness=(int)4321, signed=(boolean)true, > width=(int)16, depth=(int)16, rate=(int)24000, channels=(int)1, > channel-positions=(GstAudioChannelPosition)< > GST_AUDIO_CHANNEL_POSITION_NONE > > /GstPipeline:pipeline0/GstRtpL16Depay:rtpl16depay0.GstPad:sink: caps = > application/x-rtp, media=(string)audio, clock-rate=(int)24000, > encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, > channel-positions=(int)1, payload=(int)96, ssrc=(guint)1168011267, > clock-base=(guint)309599748, seqnum-base=(guint)27324 > /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps > = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, > width=(int)32, depth=(int)32, rate=(int)24000, channels=(int)1, > channel-positions=(GstAudioChannelPosition)< > GST_AUDIO_CHANNEL_POSITION_NONE > > /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps > = audio/x-raw-int, endianness=(int)4321, signed=(boolean)true, > width=(int)16, depth=(int)16, rate=(int)24000, channels=(int)1, > channel-positions=(GstAudioChannelPosition)< > GST_AUDIO_CHANNEL_POSITION_NONE > > /GstPipeline:pipeline0/GstAlsaSink:alsasink0.GstPad:sink: caps = > audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, > width=(int)32, depth=(int)32, rate=(int)24000, channels=(int)1, > channel-positions=(GstAudioChannelPosition)< > GST_AUDIO_CHANNEL_POSITION_NONE > > > I couldn't find a good definition of what the > GST_AUDIO_CHANNEL_POSITION_NONE means, and I wasn't able to set it to > GST_AUDIO_CHANNEL_POSITION_MONO from the command line (it looks like I > need to use python or C APIs to do that), but from what I can tell, > this comes from multichannel support, and I have just specified a > single audio channel. ?Is this what is preventing me from hearing the > audio? ?If I capture to a WAV file, and then play it back afterwards, > it sounds fine. ?So I'm not sure why streaming is failing so badly. > The playback process also dies with a "Terminated" message within a > minute or two. > > The ARM boards are running gstreamer-0.10.22. ?I see that there's been > a couple revisions since then, and if I have to upgrade to make it > work, I will. ?But I'd rather see if there's a way I can make this > version work. > > --Ron > > ------------------------------------------------------------------------ > ------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and > stay > ahead of the curve. Join us from November 9-12, 2009. Register > now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From katcipis at inf.ufsc.br Thu Sep 24 18:00:12 2009 From: katcipis at inf.ufsc.br (Tiago Katcipis) Date: Thu, 24 Sep 2009 13:00:12 -0300 Subject: [gst-devel] Noise remover In-Reply-To: <60a9403b0909231358i6497d6avc1e91d11a88d17ee@mail.gmail.com> References: <60a9403b0909231222u17d80973x7614e611bf08cd2c@mail.gmail.com> <4ABA87D0.6030103@hora-obscura.de> <60a9403b0909231358i6497d6avc1e91d11a88d17ee@mail.gmail.com> Message-ID: <60a9403b0909240900y5a7be249u6ecdb6769d65ed0a@mail.gmail.com> On Wed, Sep 23, 2009 at 5:58 PM, Tiago Katcipis wrote: > > > On Wed, Sep 23, 2009 at 5:40 PM, Stefan Kost wrote: > >> Tiago Katcipis schrieb: >> > Is there any plugin that does noise removal on an audio stream? i >> searched >> > on gstreamer site and at gst-inspect but the only noise remover that i >> have >> > found is for v?deo. >> >> We don't have one in the gst-plugins- sets. Also the algorithm depends a >> bit >> what kind of noise you need to get rid of. There is a programm called >> GnomeWaveCleaner that has some algorithms that might be good candidates >> for >> gstreamer plugins. You could also check if you find a ladspa or lv2 plugin >> and >> use that via the bridges. >> >> Stefan >> > > Thanks Stefan, ill take a look. > So i did take a look :-), and i have found Speex preprocessor too, it seens to have noise supression (even other features can be implemented using speex, like VAD and AGC). What about a plugin using speex to do general preprocessing? (AGC, Noise suppression, residual echo suppression). Im also studying the Gnome Wave Cleaner code. best regards, Katcipis > > >> >> > >> > best regards, >> > Katcipis >> > >> > >> > >> > ------------------------------------------------------------------------ >> > >> > >> ------------------------------------------------------------------------------ >> > Come build with us! The BlackBerry® Developer Conference in SF, CA >> > is the only developer event you need to attend this year. Jumpstart your >> > developing skills, take BlackBerry mobile applications to market and >> stay >> > ahead of the curve. Join us from November 9-12, 2009. Register >> now! >> > http://p.sf.net/sfu/devconf >> > >> > >> > ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > gstreamer-devel mailing list >> > gstreamer-devel at lists.sourceforge.net >> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> ------------------------------------------------------------------------------ >> Come build with us! The BlackBerry® Developer Conference in SF, CA >> is the only developer event you need to attend this year. Jumpstart your >> developing skills, take BlackBerry mobile applications to market and stay >> ahead of the curve. Join us from November 9-12, 2009. Register >> now! >> http://p.sf.net/sfu/devconf >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > > > -- > "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta > sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em menos > de 5 minutos aparecer? algu?m pra discordar de sua escolha de distribui??o, > do particionamento, do gerenciador de janelas, do ambiente de desktop, do > editor de textos..." > -- "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em menos de 5 minutos aparecer? algu?m pra discordar de sua escolha de distribui??o, do particionamento, do gerenciador de janelas, do ambiente de desktop, do editor de textos..." -------------- next part -------------- An HTML attachment was scrubbed... URL: From cmaiku at gmail.com Thu Sep 24 20:06:43 2009 From: cmaiku at gmail.com (Maiku) Date: Thu, 24 Sep 2009 13:06:43 -0500 Subject: [gst-devel] Noise remover In-Reply-To: <60a9403b0909240900y5a7be249u6ecdb6769d65ed0a@mail.gmail.com> References: <60a9403b0909231222u17d80973x7614e611bf08cd2c@mail.gmail.com> <4ABA87D0.6030103@hora-obscura.de> <60a9403b0909231358i6497d6avc1e91d11a88d17ee@mail.gmail.com> <60a9403b0909240900y5a7be249u6ecdb6769d65ed0a@mail.gmail.com> Message-ID: On Thu, Sep 24, 2009 at 11:00 AM, Tiago Katcipis wrote: > > > On Wed, Sep 23, 2009 at 5:58 PM, Tiago Katcipis > wrote: >> >> >> On Wed, Sep 23, 2009 at 5:40 PM, Stefan Kost >> wrote: >>> >>> Tiago Katcipis schrieb: >>> > Is there any plugin that does noise removal on an audio stream? i >>> > searched >>> > on gstreamer site and at gst-inspect but the only noise remover that i >>> > have >>> > found is for v?deo. >>> >>> We don't have one in the gst-plugins- sets. Also the algorithm depends a >>> bit >>> what kind of noise you need to get rid of. There is a programm called >>> GnomeWaveCleaner that has some algorithms that might be good candidates >>> for >>> gstreamer plugins. You could also check if you find a ladspa or lv2 >>> plugin and >>> use that via the bridges. >>> >>> Stefan >> >> Thanks Stefan, ill take a look. > > So i did take a look :-), and i have found Speex preprocessor too, it seens > to have noise supression (even other features can be implemented using > speex, like VAD and AGC). > > What about a plugin using speex to do general preprocessing? (AGC, Noise > suppression, residual echo suppression). > > Im also studying the Gnome Wave Cleaner code. > > best regards, > Katcipis > >> >> >>> >>> > >>> > best regards, >>> > Katcipis >>> > >>> > >>> > >>> > >>> > ------------------------------------------------------------------------ >>> > >>> > >>> > ------------------------------------------------------------------------------ >>> > Come build with us! The BlackBerry® Developer Conference in SF, CA >>> > is the only developer event you need to attend this year. Jumpstart >>> > your >>> > developing skills, take BlackBerry mobile applications to market and >>> > stay >>> > ahead of the curve. Join us from November 9-12, 2009. Register >>> > now! >>> > http://p.sf.net/sfu/devconf >>> > >>> > >>> > >>> > ------------------------------------------------------------------------ >>> > >>> > _______________________________________________ >>> > gstreamer-devel mailing list >>> > gstreamer-devel at lists.sourceforge.net >>> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Come build with us! The BlackBerry® Developer Conference in SF, CA >>> is the only developer event you need to attend this year. Jumpstart your >>> developing skills, take BlackBerry mobile applications to market and stay >>> ahead of the curve. Join us from November 9-12, 2009. Register >>> now! >>> http://p.sf.net/sfu/devconf >>> _______________________________________________ >>> gstreamer-devel mailing list >>> gstreamer-devel at lists.sourceforge.net >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> -- >> "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta >> sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em menos >> de 5 minutos aparecer? algu?m pra discordar de sua escolha de distribui??o, >> do particionamento, do gerenciador de janelas, do ambiente de desktop, do >> editor de textos..." > > > > -- > "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta > sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em menos > de 5 minutos aparecer? algu?m pra discordar de sua escolha de distribui??o, > do particionamento, do gerenciador de janelas, do ambiente de desktop, do > editor de textos..." > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > Just wanted to mention that Justing Karneges has written a speexdsp GStreamer plugin (which was started by Tester). https://delta.affinix.com/svn/trunk/psimedia/gstprovider/gstelements/speexdsp/ -Maiku From katcipis at inf.ufsc.br Thu Sep 24 23:01:37 2009 From: katcipis at inf.ufsc.br (Tiago Katcipis) Date: Thu, 24 Sep 2009 18:01:37 -0300 Subject: [gst-devel] Noise remover In-Reply-To: References: <60a9403b0909231222u17d80973x7614e611bf08cd2c@mail.gmail.com> <4ABA87D0.6030103@hora-obscura.de> <60a9403b0909231358i6497d6avc1e91d11a88d17ee@mail.gmail.com> <60a9403b0909240900y5a7be249u6ecdb6769d65ed0a@mail.gmail.com> Message-ID: <60a9403b0909241401w35171828nc09c9ab54cf4fe40@mail.gmail.com> Is this speexdsp in some gstreamer-plugin-set? i have only found speexenc and speexdec On Thu, Sep 24, 2009 at 3:06 PM, Maiku wrote: > On Thu, Sep 24, 2009 at 11:00 AM, Tiago Katcipis > wrote: > > > > > > On Wed, Sep 23, 2009 at 5:58 PM, Tiago Katcipis > > wrote: > >> > >> > >> On Wed, Sep 23, 2009 at 5:40 PM, Stefan Kost > >> wrote: > >>> > >>> Tiago Katcipis schrieb: > >>> > Is there any plugin that does noise removal on an audio stream? i > >>> > searched > >>> > on gstreamer site and at gst-inspect but the only noise remover that > i > >>> > have > >>> > found is for v?deo. > >>> > >>> We don't have one in the gst-plugins- sets. Also the algorithm depends > a > >>> bit > >>> what kind of noise you need to get rid of. There is a programm called > >>> GnomeWaveCleaner that has some algorithms that might be good candidates > >>> for > >>> gstreamer plugins. You could also check if you find a ladspa or lv2 > >>> plugin and > >>> use that via the bridges. > >>> > >>> Stefan > >> > >> Thanks Stefan, ill take a look. > > > > So i did take a look :-), and i have found Speex preprocessor too, it > seens > > to have noise supression (even other features can be implemented using > > speex, like VAD and AGC). > > > > What about a plugin using speex to do general preprocessing? (AGC, Noise > > suppression, residual echo suppression). > > > > Im also studying the Gnome Wave Cleaner code. > > > > best regards, > > Katcipis > > > >> > >> > >>> > >>> > > >>> > best regards, > >>> > Katcipis > >>> > > >>> > > >>> > > >>> > > >>> > > ------------------------------------------------------------------------ > >>> > > >>> > > >>> > > ------------------------------------------------------------------------------ > >>> > Come build with us! The BlackBerry® Developer Conference in SF, > CA > >>> > is the only developer event you need to attend this year. Jumpstart > >>> > your > >>> > developing skills, take BlackBerry mobile applications to market and > >>> > stay > >>> > ahead of the curve. Join us from November 9-12, 2009. Register > >>> > now! > >>> > http://p.sf.net/sfu/devconf > >>> > > >>> > > >>> > > >>> > > ------------------------------------------------------------------------ > >>> > > >>> > _______________________________________________ > >>> > gstreamer-devel mailing list > >>> > gstreamer-devel at lists.sourceforge.net > >>> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > >>> > >>> > >>> > >>> > ------------------------------------------------------------------------------ > >>> Come build with us! The BlackBerry® Developer Conference in SF, CA > >>> is the only developer event you need to attend this year. Jumpstart > your > >>> developing skills, take BlackBerry mobile applications to market and > stay > >>> ahead of the curve. Join us from November 9-12, 2009. Register > >>> now! > >>> http://p.sf.net/sfu/devconf > >>> _______________________________________________ > >>> gstreamer-devel mailing list > >>> gstreamer-devel at lists.sourceforge.net > >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > >> > >> > >> > >> -- > >> "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta > >> sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em > menos > >> de 5 minutos aparecer? algu?m pra discordar de sua escolha de > distribui??o, > >> do particionamento, do gerenciador de janelas, do ambiente de desktop, > do > >> editor de textos..." > > > > > > > > -- > > "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta > > sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em > menos > > de 5 minutos aparecer? algu?m pra discordar de sua escolha de > distribui??o, > > do particionamento, do gerenciador de janelas, do ambiente de desktop, do > > editor de textos..." > > > > > ------------------------------------------------------------------------------ > > Come build with us! The BlackBerry® Developer Conference in SF, CA > > is the only developer event you need to attend this year. Jumpstart your > > developing skills, take BlackBerry mobile applications to market and stay > > ahead of the curve. Join us from November 9-12, 2009. Register > now! > > http://p.sf.net/sfu/devconf > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > Just wanted to mention that Justing Karneges has written a speexdsp > GStreamer plugin (which was started by Tester). > > https://delta.affinix.com/svn/trunk/psimedia/gstprovider/gstelements/speexdsp/ > > -Maiku > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em menos de 5 minutos aparecer? algu?m pra discordar de sua escolha de distribui??o, do particionamento, do gerenciador de janelas, do ambiente de desktop, do editor de textos..." -------------- next part -------------- An HTML attachment was scrubbed... URL: From lists at svrinformatica.it Fri Sep 25 01:11:49 2009 From: lists at svrinformatica.it (Mailing List SVR) Date: Fri, 25 Sep 2009 01:11:49 +0200 Subject: [gst-devel] multiudpsink fd leak, BUG? Message-ID: <200909250111.49632.lists@svrinformatica.it> Hi all, I have a file descriptor leak, the problem arise when I use the remove signal in multiudpsink. To add a new client to multiudpsink in my python code I do something like this: self.multisink.emit('add',host,port) and to remove a client I use: self.multisink.emit('remove',host,port) 1) when I add a new client two fd are used 2) when I remove a client the file descriptor aren't given back 3) if I stop the whole stream only the file descriptor for the current connected client are given back so if in a for loop I add and remove client to a multiudpsink element I have a file descriptor leak, is there something wrong in my code or this is a gstreamer bug? thanks Nicola From josel.segura at gmail.com Fri Sep 25 09:29:44 2009 From: josel.segura at gmail.com (=?ISO-8859-1?Q?Jos=E9_Luis?=) Date: Fri, 25 Sep 2009 09:29:44 +0200 Subject: [gst-devel] Using rtspsrc Message-ID: <2de9bf90909250029j979f747q12d38b97000df7e1@mail.gmail.com> Hi! This is my first mail to this list, and I'm using Gstreamer directly on C only few days ago. I'm trying to use the rtspsrc element to show a RTSP stream from a IP camera. On my first try, I used gst-launch tool to see that it's possible to construct this pipeline: $ gst-launch-0.10 rtspsrc location="uri_of_my_cam" ! decodebin ! xvimagesink and it works fine! But, when I tried the same inside a program... I create the 3 elements, add it to the pipeline and then I linked it between. It hink my problem was the use of decodebin, but I read some examples on the documentation and I'm using it in the right way. When I output debug messages from the bus, I saw something strange: gstreamer notify me that rtspsrc is not properly connected to decodebin. I read some documentation about rtspsrc and I found that it's a GstBin, not a simply GstElement. Inside the rtspsrc bin would be a source pad, but I'm not sure how can I use it. Is there any signal that notifies me about the creation of the source pad? Have I ask to rtspsrc for a source pad? I had searched on Google for use-examples of rtspsrc, but I can't find anything related. Actually, I'm using "playbin" to play the RTSP camera, but I want to learn this kind of things! Thanks for your attention and excuse me, I'm not very good at English. -------------- next part -------------- An HTML attachment was scrubbed... URL: From gstelzz at yahoo.fr Fri Sep 25 09:51:33 2009 From: gstelzz at yahoo.fr (Aurelien Grimaud) Date: Fri, 25 Sep 2009 09:51:33 +0200 Subject: [gst-devel] Using rtspsrc In-Reply-To: <2de9bf90909250029j979f747q12d38b97000df7e1@mail.gmail.com> References: <2de9bf90909250029j979f747q12d38b97000df7e1@mail.gmail.com> Message-ID: <4ABC7685.7080805@yahoo.fr> Hi, rtspsrc source pads are "sometimes" pad. Therefore you have to connect a GstElement : "pad-added" signal on rtspsrc, and link downstream in call back. Aurelien Jos? Luis a ?crit : > Hi! > > This is my first mail to this list, and I'm using Gstreamer directly > on C only few days ago. > > I'm trying to use the rtspsrc element to show a RTSP stream from a IP > camera. On my first try, I used gst-launch tool to see that it's > possible to construct this pipeline: > > $ gst-launch-0.10 rtspsrc location="uri_of_my_cam" ! decodebin ! > xvimagesink > > and it works fine! But, when I tried the same inside a program... I > create the 3 elements, add it to the pipeline and then I linked it > between. > > It hink my problem was the use of decodebin, but I read some examples > on the documentation and I'm using it in the right way. > > When I output debug messages from the bus, I saw something strange: > gstreamer notify me that rtspsrc is not properly connected to > decodebin. I read some documentation about rtspsrc and I found that > it's a GstBin, not a simply GstElement. > > Inside the rtspsrc bin would be a source pad, but I'm not sure how can > I use it. Is there any signal that notifies me about the creation of > the source pad? Have I ask to rtspsrc for a source pad? > > I had searched on Google for use-examples of rtspsrc, but I can't find > anything related. > > Actually, I'm using "playbin" to play the RTSP camera, but I want to > learn this kind of things! > > Thanks for your attention and excuse me, I'm not very good at English. > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rohan at perzonae.com Fri Sep 25 10:36:54 2009 From: rohan at perzonae.com (Rohan) Date: Fri, 25 Sep 2009 10:36:54 +0200 Subject: [gst-devel] creating bins and ghostpads In-Reply-To: <4ABB3BD9.7090206@yahoo.fr> References: <4ABB38CD.902@perzonae.com> <4ABB3BD9.7090206@yahoo.fr> Message-ID: <4ABC8126.3050809@perzonae.com> Aurelien Grimaud wrote: > Add vidsource in pipeline ? > Oh lord, how embarrassing. Thanks, that is exactly what the problem was. Thanks again, Rohan > Rohan a ?crit : >> Hi all, >> >> I am obviously doing something wrong here, but cannot figure out >> what. >> >> Essentially the pipeline is the same as the receiver but I have broken >> it into bins. I seem to be having problems with the ghostpad, because >> I get this error: >> >> (stream_receiver.py:1914): GStreamer-WARNING **: Trying to connect elements that >> don't share a common ancestor: vidsource and vidbin >> >> Traceback (most recent call last): >> File "stream_receiver.py", line 118, in >> server() >> File "stream_receiver.py", line 27, in __init__ >> vidsource.link(vidbin) >> gst.LinkError: failed to link vidsource with vidbin >> >> Here is the code: >> >> ---------------------------------------------------------------------- >> #!/bin/env python >> >> # The gst pipeline >> # pipe="udpsrc port=5000 ! smokedec ! autovideosink" >> # # audio >> # pipe="$pipe tcpclientsrc host=127.0.0.1 port=5001 ! " >> # pipe="$pipe speexdec ! queue ! alsasink sync=false " >> >> import sys,os >> import gobject >> import pygst >> pygst.require("0.10") >> import gst >> >> class server (object): >> def __init__(self): >> # start with video >> >> self.pipe = gst.Pipeline("player") >> >> # video source >> vidsource = gst.element_factory_make("udpsrc", "vidsource") >> vidsource.set_property("port", 5000) >> >> vidbin = self.buildvid() >> self.pipe.add(vidbin) >> vidsource.link(vidbin) >> >> >> self.pipe.set_state(gst.STATE_PLAYING) >> >> def buildvid(self): >> bin = gst.Bin("vidbin") >> queue = gst.element_factory_make("queue") >> smokedec = gst.element_factory_make("smokedec") >> vidsink = gst.element_factory_make("autovideosink") >> bin.add(queue, smokedec, vidsink) >> gst.element_link_many(queue, smokedec, vidsink) >> >> # ghostpad >> binsink = gst.GhostPad("binsink", queue.get_pad("sink")) >> bin.add_pad(binsink) >> return bin >> >> if __name__ == "__main__": >> server() >> loop = gobject.MainLoop() >> loop.run() >> >> ---------------------------------------------------------------------------- >> >> What is driving me slightly nuts is that if I put all the buildvid code >> into init and have a long series of object instatiations with add to >> pipeline and link it works, but as soon as I started messing with >> ghostpads and bin (which makes the code much more manageable when >> adding functionality for windows and macs) this messed up. Below is >> more code that works, and I cannot figure out why this and why not >> that. >> >> And the other thing is this does work in a commandline pipeline such >> as the one sitting at the top of the script in comments. >> >> I am using the same system for the sender, but I'll only include the >> local display. This sends an image happily, using the build_localvid >> bin, and linking it to the camera. >> >> ----------------------------------------------------------------------------- >> #!/bin/env python >> >> import sys, os >> import gobject >> import pygst >> pygst.require("0.10") >> import gst >> >> class client(object): >> >> def __init__(self): >> self.pipe = gst.Pipeline("sender") >> >> # Initial video input >> camera = gst.element_factory_make("v4l2src", "camera") >> vidtee = gst.element_factory_make("tee", "vidtee") >> self.pipe.add(camera, vidtee) >> camera.link(vidtee) >> >> # local video >> localvidbin = self.build_localvid() >> self.pipe.add(localvidbin) >> vidtee.link(localvidbin) >> >> self.pipe.set_state(gst.STATE_PLAYING) >> >> def build_localvid(self): >> """This bin takes a camera (video) stream and produces a live image >> locally.""" >> bin = gst.Bin("localvid") >> queue = gst.element_factory_make("queue") >> out = gst.element_factory_make("xvimagesink") >> bin.add(out, queue) >> queue.link_pads("src", out, "sink") >> binsink = gst.GhostPad("binsink", queue.get_pad("sink")) >> bin.add_pad(binsink) >> return bin >> >> if __name__ == '__main__': >> client() >> loop = gobject.MainLoop() >> loop.run() >> >> --------------------------------------------------------------------------- >> >> This does the correct thing, and I get a live image from the camera. >> >> I am completely stumped on this one, and have resorted to using >> gst.parse_bin_from_description() with the gst-launch pipeline, but >> this is far from ideal from a maintenance perspective. :) >> >> I am sure I am missing something pretty obvious, but so far flailing >> around in my ignorance has not let me stumble on a solution. >> >> Thanks for any help, >> >> Rohan >> >> >> ------------------------------------------------------------------------------ >> Come build with us! The BlackBerry® Developer Conference in SF, CA >> is the only developer event you need to attend this year. Jumpstart your >> developing skills, take BlackBerry mobile applications to market and stay >> ahead of the curve. Join us from November 9-12, 2009. Register now! >> http://p.sf.net/sfu/devconf >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From dxssx.dxssx at gmail.com Fri Sep 25 12:06:32 2009 From: dxssx.dxssx at gmail.com (dxssx) Date: Fri, 25 Sep 2009 18:06:32 +0800 Subject: [gst-devel] how can I make gst-rtsp-server as a VOD server? Message-ID: Hi, I want to use gst-rtsp-server as my vod server, but I found it is too hard for some reasons: 1. rtsp-server has no flow control in sending packet. It sends packets as fast as possible which makes client can not decode in time. 2. rtsp-server has no dynamic url. Everytime I add a file into vod list, I need to add a url mapping for it. 3. rtsp-server doesn't support dynamic pipeline. Everytime I add a file into vod list, I need to rewrite a new pipeline to parse and rtppay it. I am new here, so I am afraid that I missed something useful in rtsp-server or gstreamer to overcome the problems above. Does anyone can help me? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: From optavas at gmail.com Fri Sep 25 20:37:46 2009 From: optavas at gmail.com (Ottaviano Vasselli) Date: Fri, 25 Sep 2009 20:37:46 +0200 Subject: [gst-devel] unable to load pipeline from XML Message-ID: <6c9581cb0909251137v15601b0m5bfede30a2f181a@mail.gmail.com> Hello, I'm having trouble with gst-xmllaunch. When I launch that simple pipeline with gst-launch, everything goes fine: gst-launch-0.10 filesrc location=$SRC_FILE ! decodebin name=decode decode. \ ! queue ! theoraenc ! oggmux name=mux decode. ! queue ! audioconvert ! vorbisenc \ ! mux. mux. ! filesink location=$DST_FILE (with $SRC_FILE and $DST_FILE existing file names) so I added "-o saved_pipe.xml" obtaining an xml file looking valid. But when I run gst-xmllaunch -vvv saved_pipe.xml , transcoding does not start and pipeline terminates with following output: (gst-launch-0.10:3746): GStreamer-CRITICAL **: pad sink_563092733 is not a source pad (gst-launch-0.10:3746): GStreamer-WARNING **: Name fakesink is not unique in bin decode, not adding (gst-launch-0.10:3746): GStreamer-WARNING **: Name typefind is not unique in bin decode, not adding francesco at helium:/tmp$ gst-xmllaunch-0.10 -vvv test.xml (gst-launch-0.10:3758): GStreamer-CRITICAL **: pad sink_563092733 is not a source pad (gst-launch-0.10:3758): GStreamer-WARNING **: Name fakesink is not unique in bin decode, not adding (gst-launch-0.10:3758): GStreamer-WARNING **: Name typefind is not unique in bin decode, not adding (gst-launch-0.10:3758): GStreamer-CRITICAL **: gst_pad_load_and_link: assertion `pad != NULL' failed Setting pipeline to PAUSED ... /pipeline0/decode/typefind.src: caps = video/x-msvideo Pipeline is PREROLLING ... ... Also tried adding filesrc0.location="..." filesink0.location="..." with real files specified, but no change. Does somebody have any clue about? Thank you for your attention, bye From marcelomendozasc at gmail.com Fri Sep 25 21:15:22 2009 From: marcelomendozasc at gmail.com (=?ISO-8859-1?Q?Marcelo_de_S=E1_Mendoza?=) Date: Fri, 25 Sep 2009 16:15:22 -0300 Subject: [gst-devel] GSM Plugins gsmdec gsmenc under Cross-Compiling i586-mingw32msvc Message-ID: <3d2d37660909251215n30a46b56sfbc60201ca2ae3ac@mail.gmail.com> Hey all I'm trying to cross-compile gsm plugins but no sucess. When I compile for linux works fine. But crossing i586-mingw32msvc under linux is not working...I noticed that the configure searchs for the gsm.h file but in the gstpluginsbad there's no gsm.h file...any tip please? =/ I'm kind of new on compiling libs process. Thanks! My configure parameters is: ./configure --prefix=/usr/local/mingw32 --build=i686-pc-linux-gnu --host=i586-mingw32msvc --disable-acm --enable-external=yes --enable-directdraw=yes --enable-directsound=yes --enable-gsm=yes --enable-dependency-tracking=yes --enable-shared=yes --disable-apexsink And failure log of configure is: configure: *** checking feature: GSM library *** configure: *** for plug-ins: gsmenc gsmdec *** checking for gsm_create in -lgsm... no checking for gsm_create in -lgsm... (cached) no configure: *** These plugins will not be built: gsmenc gsmdec -------------- next part -------------- An HTML attachment was scrubbed... URL: From daltoncezane at gmail.com Fri Sep 25 21:24:30 2009 From: daltoncezane at gmail.com (=?ISO-8859-1?Q?Dalton_C=E9zane?=) Date: Fri, 25 Sep 2009 16:24:30 -0300 Subject: [gst-devel] gstreamer-devel Digest, Vol 40, Issue 30 In-Reply-To: References: Message-ID: <98029fd50909251224h4d98f92boe11ff1fdc4d5170f@mail.gmail.com> > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sat, 19 Sep 2009 11:10:44 +0200 > From: Edward Hervey > Subject: Re: [gst-devel] Demux H264 video and HE-AAC audio > To: Discussion of the development of GStreamer > > Message-ID: <1253351444.8398.0.camel at putamadre> > Content-Type: text/plain; charset="UTF-8" > > Use mpegtsdemux and not ffdemux_mpegts. The rule of thumb is : don't use > the ffmpeg demuxers (they have a rank of NONE for a reason). > > Edward > Hi, Edward. Thanks for your answer. But, I could not install the mpegtsdemux. I saw that gstreamer-plugins-bad contains it but when I use "sudo apt-get install gstreamer0.10-plugins-bad" and nothing (I am using Ubuntu 8.10)... the packet "bad" is already the newer here in my OS. Can you tell me how I install this plugin? And, Sean (below), the problem is not with osssink. I tried to run with the alsasink too, but it did not work. The error is: gst-launch-0.10 filesrc location="GLOBO188bytes.mpg" ! ffdemux_mpegts name=d d. queue ! ffdec_h264 ! xvimagesink d. queue ! faad ! audioconvert ! alsasink Leaving the connection as PAUSED ... The connection is making PREROLL ... ERROR: of the element /GstPipeline:pipeline0/ffdemux_mpegts:d: Internal data stream error. Aditional debbuging Information: gstffmpegdemux.c(1364): gst_ffmpegdemux_loop (): /GstPipeline:pipeline0/ffdemux_mpegts:d: streaming stopped, reason error ERROR: the connection does not want to make preroll. Leaving the connection in NULL ... FREEING the connection ... Thanks in advance! > > Message: 2 > Date: Sat, 19 Sep 2009 11:47:00 -0400 > From: Sean McNamara > Subject: Re: [gst-devel] Demux H264 video and HE-AAC audio > To: Discussion of the development of GStreamer > > Message-ID: > <74eb1fe20909190847r4f1c8414m95e9bd3cec251b1b at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > Are you sure the problem isn't with osssink? Post the entire output of > using gst-launch with this command, and you may also try autoaudiosink > instead of osssink. If you're sure the problem isn't with the > audiosink, I'll need to see the rest of the output to know what > exactly went wrong. > > HTH, > > Sean > -- ======================================================= Dalton C?zane - Voip UFCG: 1075-2005 Mestrando em Ci?ncia da Computa??o (UFCG) Bacharel em Ci?ncia da Computa??o (UFCG) T?cnico em Inform?tica (ETER) -------------- next part -------------- An HTML attachment was scrubbed... URL: From daltoncezane at gmail.com Fri Sep 25 23:49:08 2009 From: daltoncezane at gmail.com (=?ISO-8859-1?Q?Dalton_C=E9zane?=) Date: Fri, 25 Sep 2009 18:49:08 -0300 Subject: [gst-devel] Demux H264 video and HE-AAC audio (Edward Hervey) Message-ID: <98029fd50909251449w6d13ee21kab5657baf83aab00@mail.gmail.com> I tried to run h264 video and he-aac audio in another machine with mpegtsdemux, but it did not work: /home/dalton# gst-launch-0.10 filesrc location="GLOBO188bytes.mpg" ! mpegtsdemux name=demux demux. queue ! ffdec_h264 ! xvimagesink demux. queue ! faad ! audioconvert ! alsasink Setting pipeline to PAUSED ... Pipeline is PREROLLING ... ..... And continues like it was in loop... Can anyone help me? _____________________________________________________________________________________________________ Use mpegtsdemux and not ffdemux_mpegts. The rule of thumb is : don't use the ffmpeg demuxers (they have a rank of NONE for a reason). Edward On Fri, 2009-09-18 at 16:42 -0300, Dalton C?zane wrote: > Hi all, > I am new at list and at the GStreamer study. I am trying to demux H264 > video and HE-AAC audio with gst-launch-0.10. > I already succeeded just the video, without audio, with this command > line: gst-launch-0.10 filesrc location="GLOBO188bytes.mpg" ! > ffdemux_mpegts name=demux demux. queue ! ffdec_h264 ! xvimagesink ! > demux. queue ! faad ! audioconvert ! osssink > > This way, the video is displayed but the sound does not play. > Can anyone help me? Some tip? > > Thanks in advance. -- ======================================================= Dalton C?zane - Voip UFCG: 1075-2005 Mestrando em Ci?ncia da Computa??o (UFCG) Bacharel em Ci?ncia da Computa??o (UFCG) T?cnico em Inform?tica (ETER) -------------- next part -------------- An HTML attachment was scrubbed... URL: From cmaiku at gmail.com Sat Sep 26 03:08:23 2009 From: cmaiku at gmail.com (Maiku) Date: Fri, 25 Sep 2009 20:08:23 -0500 Subject: [gst-devel] Noise remover In-Reply-To: <60a9403b0909241401w35171828nc09c9ab54cf4fe40@mail.gmail.com> References: <60a9403b0909231222u17d80973x7614e611bf08cd2c@mail.gmail.com> <4ABA87D0.6030103@hora-obscura.de> <60a9403b0909231358i6497d6avc1e91d11a88d17ee@mail.gmail.com> <60a9403b0909240900y5a7be249u6ecdb6769d65ed0a@mail.gmail.com> <60a9403b0909241401w35171828nc09c9ab54cf4fe40@mail.gmail.com> Message-ID: On Thu, Sep 24, 2009 at 4:01 PM, Tiago Katcipis wrote: > Is this speexdsp in some gstreamer-plugin-set? i have only found speexenc > and speexdec > No. Justin says it still needs to be cleaned up before being added into any plugin package. -Maiku From katcipis at inf.ufsc.br Sat Sep 26 13:14:29 2009 From: katcipis at inf.ufsc.br (Tiago Katcipis) Date: Sat, 26 Sep 2009 08:14:29 -0300 Subject: [gst-devel] Noise remover In-Reply-To: References: <60a9403b0909231222u17d80973x7614e611bf08cd2c@mail.gmail.com> <4ABA87D0.6030103@hora-obscura.de> <60a9403b0909231358i6497d6avc1e91d11a88d17ee@mail.gmail.com> <60a9403b0909240900y5a7be249u6ecdb6769d65ed0a@mail.gmail.com> <60a9403b0909241401w35171828nc09c9ab54cf4fe40@mail.gmail.com> Message-ID: <60a9403b0909260414p133b689fy33e3a113c88c58b3@mail.gmail.com> if it already works and only needs to be cleaned isnt better to add it to gst-plugins-bad? So everyone will know that it exists, will be able to use it and to clean it too. best regards, Katcipis On Fri, Sep 25, 2009 at 10:08 PM, Maiku wrote: > On Thu, Sep 24, 2009 at 4:01 PM, Tiago Katcipis > wrote: > > Is this speexdsp in some gstreamer-plugin-set? i have only found speexenc > > and speexdec > > > > No. Justin says it still needs to be cleaned up before being added > into any plugin package. > > -Maiku > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em menos de 5 minutos aparecer? algu?m pra discordar de sua escolha de distribui??o, do particionamento, do gerenciador de janelas, do ambiente de desktop, do editor de textos..." -------------- next part -------------- An HTML attachment was scrubbed... URL: From wim.taymans at gmail.com Sat Sep 26 15:28:29 2009 From: wim.taymans at gmail.com (Wim Taymans) Date: Sat, 26 Sep 2009 15:28:29 +0200 Subject: [gst-devel] how can I make gst-rtsp-server as a VOD server? In-Reply-To: References: Message-ID: <1253971709.4654.18.camel@metal> On Fri, 2009-09-25 at 18:06 +0800, dxssx wrote: > Hi, > I want to use gst-rtsp-server as my vod server, but I found it is too > hard for some reasons: > > > 1. rtsp-server has no flow control in sending packet. > It sends packets as fast as possible which makes client can not decode > in time. for flow control if relies on a gstreamer element that produces correct timestamps. If you are, for example, sending an mp3 file, you need to insert mp3parse after the filesrc so that timestamps are properly set on the data. > > > 2. rtsp-server has no dynamic url. > Everytime I add a file into vod list, I need to add a url mapping for > it. You can make a subclass of the mediafactory to dynamically create pipelines. The example base class only accept gst-launch type syntax. > > > 3. rtsp-server doesn't support dynamic pipeline. > Everytime I add a file into vod list, I need to rewrite a new pipeline > to parse and rtppay it. Again this can be implemented using subclasses of the standard example factories. Wim > > > I am new here, so I am afraid that I missed something useful in > rtsp-server or gstreamer to overcome the problems above. > > > Does anyone can help me? > > > Thanks. > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From t.i.m at zen.co.uk Sat Sep 26 22:28:45 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Sat, 26 Sep 2009 21:28:45 +0100 Subject: [gst-devel] [gst-cvs] gst-plugins-bad: videosignal: change pattern data type to uint64, add property and message field In-Reply-To: <4ABE716F.2070601@hora-obscura.de> References: <20090926161526.AD63010050@kemper.freedesktop.org> <4ABE716F.2070601@hora-obscura.de> Message-ID: <1253996925.11264.21.camel@zingle> On Sat, 2009-09-26 at 22:54 +0300, Stefan Kost wrote: > > videosignal: change pattern data type to uint64, add property and message field > > > > Keeps the old uint typed value support for compatibility. > > > > This element is in plugins bad. I think its fine to just break the API. If > people use this for real, then they should push that it goes to good :) Well, you're right of course, but for elements where we know that people are using them already and where it costs us hardly anything to keep things backwards compatible, we should do that IMO. We can then remove the deprecated API after a while or when the element is moved to -ugly or -good. And we really shouldn't ever change API in a way that may lead to crashes for users of the old API (as changing a property from uint to uint64 type without changing the property name might, for example). Also, if we're not even trying to accommodate existing users of -bad elements where it's easy for us to do so, then people will just pressure us to move elements into -good prematurely. That's not something I'm particularly keen on. Cheers -Tim From yangsb05 at gmail.com Sun Sep 27 04:10:58 2009 From: yangsb05 at gmail.com (yangsb) Date: Sat, 26 Sep 2009 19:10:58 -0700 (PDT) Subject: [gst-devel] how can I make gst-rtsp-server as a VOD server? In-Reply-To: <1253971709.4654.18.camel@metal> References: <1253971709.4654.18.camel@metal> Message-ID: <25630294.post@talk.nabble.com> As far as I know , Just the same as Wim mentioned, after insert mp3parse after the filesrc , the vod for mp3 files works fine. Thanks. Best regards. Wim Taymans-2 wrote: > > On Fri, 2009-09-25 at 18:06 +0800, dxssx wrote: >> Hi, >> I want to use gst-rtsp-server as my vod server, but I found it is too >> hard for some reasons: >> >> >> 1. rtsp-server has no flow control in sending packet. >> It sends packets as fast as possible which makes client can not decode >> in time. > > for flow control if relies on a gstreamer element that produces correct > timestamps. If you are, for example, sending an mp3 file, you need to > insert mp3parse after the filesrc so that timestamps are properly set on > the data. >> >> >> 2. rtsp-server has no dynamic url. >> Everytime I add a file into vod list, I need to add a url mapping for >> it. > > You can make a subclass of the mediafactory to dynamically create > pipelines. The example base class only accept gst-launch type syntax. > >> >> >> 3. rtsp-server doesn't support dynamic pipeline. >> Everytime I add a file into vod list, I need to rewrite a new pipeline >> to parse and rtppay it. > > Again this can be implemented using subclasses of the standard example > factories. > > Wim >> >> >> I am new here, so I am afraid that I missed something useful in >> rtsp-server or gstreamer to overcome the problems above. >> >> >> Does anyone can help me? >> >> >> Thanks. >> >> >> ------------------------------------------------------------------------------ >> Come build with us! The BlackBerry® Developer Conference in SF, CA >> is the only developer event you need to attend this year. Jumpstart your >> developing skills, take BlackBerry mobile applications to market and stay >> ahead of the curve. Join us from November 9-12, 2009. Register >> now! >> http://p.sf.net/sfu/devconf >> _______________________________________________ gstreamer-devel mailing >> list gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register > now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- View this message in context: http://www.nabble.com/how-can-I-make-gst-rtsp-server-as-a-VOD-server--tp25609556p25630294.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From yangsb05 at gmail.com Sun Sep 27 05:01:34 2009 From: yangsb05 at gmail.com (yangsb) Date: Sat, 26 Sep 2009 20:01:34 -0700 (PDT) Subject: [gst-devel] How to Free resources in gst-rtsp-server ? Message-ID: <25630515.post@talk.nabble.com> Hi ,guys. I used gst-rtsp-server for several months. Now , I found that some resource can't be free correctly . I checked the code in gst-launch.c , and found that there is such code to free resources: ****************************************************** gst_element_set_state ( pipeline, GST_STATE_NULL )? ****************************************************** However , in gst-rtsp-server/examples/test-launch.c ?there isn't any such codes. So, is there any api in gst-rtsp-server that can be used to free resources ? I think this is very useful . Thanks. Best regards. -- View this message in context: http://www.nabble.com/How-to-Free-resources-in-gst-rtsp-server---tp25630515p25630515.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From mail at renestadler.de Sun Sep 27 11:45:18 2009 From: mail at renestadler.de (=?UTF-8?B?UmVuw6kgU3RhZGxlcg==?=) Date: Sun, 27 Sep 2009 12:45:18 +0300 Subject: [gst-devel] [gst-cvs] gst-plugins-bad: videosignal: change pattern data type to uint64, add property and message field In-Reply-To: <1253996925.11264.21.camel@zingle> References: <20090926161526.AD63010050@kemper.freedesktop.org> <4ABE716F.2070601@hora-obscura.de> <1253996925.11264.21.camel@zingle> Message-ID: <4ABF342E.6000307@renestadler.de> Tim-Philipp M?ller schrieb: > On Sat, 2009-09-26 at 22:54 +0300, Stefan Kost wrote: > >>> videosignal: change pattern data type to uint64, add property and message field >>> >>> Keeps the old uint typed value support for compatibility. >>> >> This element is in plugins bad. I think its fine to just break the API. If >> people use this for real, then they should push that it goes to good :) Actually, _I'm_ using it for real of course, and this is the best way of breaking the chicken-and-egg situation for me :) > Well, you're right of course, but for elements where we know that people > are using them already and where it costs us hardly anything to keep > things backwards compatible, we should do that IMO. We can then remove > the deprecated API after a while or when the element is moved to -ugly > or -good. > > And we really shouldn't ever change API in a way that may lead to > crashes for users of the old API (as changing a property from uint to > uint64 type without changing the property name might, for example). Works perfectly here since I believe the property name and message field should really keep the *-uint64 name in the long term. There's some gotchas about handling 64 bit integers in C, so I like putting a fat reminder about that right into the API. Just think g_object_set (queue, "max-level-bytes", 0, "max-level-time", 0, NULL), it looks so innocent :) > Also, if we're not even trying to accommodate existing users of -bad > elements where it's easy for us to do so, then people will just pressure > us to move elements into -good prematurely. That's not something I'm > particularly keen on. > > Cheers > -Tim -- Regards, Ren? Stadler From mvjohn100 at hotmail.com Sun Sep 27 12:59:55 2009 From: mvjohn100 at hotmail.com (john verghese) Date: Sun, 27 Sep 2009 16:29:55 +0530 Subject: [gst-devel] gstream development in window Message-ID: Hello Please can help on the followings 1) Is it possible to make play "3gp" file using gstream in windows? 2)In windows XP using gstream... the file play properly but the following messages remains. it is a serious problem? gst is my "exe" name Thanks, John ** (gst.exe:1652): CRITICAL **: file E:\devel-release\src_releases \gstreamer\gst \gstindex.c: line 601: assertion `GST_IS_INDEX (index)' failed ** (gst.exe:1652): WARNING **: Add decoder dsicinvideo (97) please ** (gst.exe:1652): WARNING **: Add decoder gif (100) please ** (gst.exe:1652): WARNING **: Add decoder kmvc (88) please ** (gst.exe:1652): WARNING **: Add decoder nuv (87) please ** (gst.exe:1652): WARNING **: Add decoder smackvid (86) please ** (gst.exe:1652): WARNING **: Add decoder targa (96) please ** (gst.exe:1652): WARNING **: Add decoder tiertexseqvideo (98) please ** (gst.exe:1652): WARNING **: Add decoder tiff (99) please ** (gst.exe:1652): WARNING **: Add decoder VMware video (92) please ** (gst.exe:1652): WARNING **: Add decoder dsicinaudio (86045) please ** (gst.exe:1652): WARNING **: Add decoder imc (86046) please ** (gst.exe:1652): WARNING **: Add decoder smackaud (86042) please ** (gst.exe:1652): WARNING **: Add decoder wavpack (86044) please OIL: ERROR liboilcpu.c 282: oil_cpu_i386_kernel_restrict_flags(): Operating syst em is not known to support SSE.? Assuming it does, which might cause problems _________________________________________________________________ cricket and news. Logon to MSN Video for the latest clips http://www.exploremyway.com From dxssx.dxssx at gmail.com Mon Sep 28 06:13:05 2009 From: dxssx.dxssx at gmail.com (dxssx) Date: Mon, 28 Sep 2009 12:13:05 +0800 Subject: [gst-devel] how can I make gst-rtsp-server as a VOD server? In-Reply-To: <1253971709.4654.18.camel@metal> References: <1253971709.4654.18.camel@metal> Message-ID: Thanks, Wim. Sorry for problem 1. My server can not control the flow because I have modified rtsp server by g_object_set (G_OBJECT (udpsink0), "sync", FALSE, NULL); When I enable the sync, it works well with mp3parse pipeline. I will test the server with h264parse, aacparse and others. For problem 2, 3: Is there any plugin can auto detect acodec/vcodec of a file? BTW: Is there a roadmap for rtsp-server? Thanks again. 2009/9/26 Wim Taymans > On Fri, 2009-09-25 at 18:06 +0800, dxssx wrote: > > Hi, > > I want to use gst-rtsp-server as my vod server, but I found it is too > > hard for some reasons: > > > > > > 1. rtsp-server has no flow control in sending packet. > > It sends packets as fast as possible which makes client can not decode > > in time. > > for flow control if relies on a gstreamer element that produces correct > timestamps. If you are, for example, sending an mp3 file, you need to > insert mp3parse after the filesrc so that timestamps are properly set on > the data. > > > > > > 2. rtsp-server has no dynamic url. > > Everytime I add a file into vod list, I need to add a url mapping for > > it. > > You can make a subclass of the mediafactory to dynamically create > pipelines. The example base class only accept gst-launch type syntax. > > > > > > > 3. rtsp-server doesn't support dynamic pipeline. > > Everytime I add a file into vod list, I need to rewrite a new pipeline > > to parse and rtppay it. > > Again this can be implemented using subclasses of the standard example > factories. > > Wim > > > > > > I am new here, so I am afraid that I missed something useful in > > rtsp-server or gstreamer to overcome the problems above. > > > > > > Does anyone can help me? > > > > > > Thanks. > > > > > > > ------------------------------------------------------------------------------ > > Come build with us! The BlackBerry® Developer Conference in SF, CA > > is the only developer event you need to attend this year. Jumpstart your > > developing skills, take BlackBerry mobile applications to market and stay > > ahead of the curve. Join us from November 9-12, 2009. Register > now! > > http://p.sf.net/sfu/devconf > > _______________________________________________ gstreamer-devel mailing > list gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From 4ernov at gmail.com Mon Sep 28 11:47:26 2009 From: 4ernov at gmail.com (Alexey Chernov) Date: Mon, 28 Sep 2009 13:47:26 +0400 Subject: [gst-devel] Problems with deinterleave plugin Message-ID: <200909281347.26913.4ernov@gmail.com> Hello, I'm working on a sound editor based on GStreamer and I faced the problem with deinterleave plugin recently. To load and decode file I use pipeline: filesrc, decodebin, audioconvert, audioresample, deinterleave and then several branches containing queue, audioconvert and appsink (for the experiment I changed it to fakesink). Here's the code of main function: GMainLoop* _loop; GstElement* _pipeline; void load_file(const char* filename) { GstElement *source, *decodebin, *audio_convert, *audio_resample, *deint; GstBus* bus; if (_loop) { g_main_loop_quit(_loop); g_main_loop_unref(_loop); } _loop = g_main_loop_new (NULL, FALSE); /* Create gstreamer elements */ if (_pipeline) { gst_element_set_state (_pipeline, GST_STATE_NULL); gst_object_unref (GST_OBJECT (_pipeline)); _pipeline=0; } _pipeline = gst_pipeline_new("decode_to_app"); source = gst_element_factory_make("filesrc", "filesrc"); decodebin = gst_element_factory_make("decodebin", "decode_bin"); audio_convert = gst_element_factory_make("audioconvert","audio-convert"); audio_resample = gst_element_factory_make("audioresample","audio-resample"); deint = gst_element_factory_make("deinterleave", "deint"); if (!_pipeline || !source || !decodebin || !audio_convert || !deint) { std::cerr<<"Elements could not be created. Exiting."< Received event on flushing pad. Discarding 0:00:01.703978717 5174 0x1d712a0 INFO GST_EVENT gstpad.c:4675:gst_pad_send_event: Received event on flushing pad. Discarding 0:00:01.703995479 5174 0x1d712a0 INFO GST_EVENT gstpad.c:4675:gst_pad_send_event: Received event on flushing pad. Discarding 0:00:01.704007213 5174 0x1d712a0 INFO GST_EVENT gstpad.c:4675:gst_pad_send_event: Received event on flushing pad. Discarding 0:00:01.704021111 5174 0x1d712a0 INFO GST_EVENT gstpad.c:4675:gst_pad_send_event: Received event on flushing pad. Discarding 0:00:01.704032565 5174 0x1d712a0 INFO GST_EVENT gstpad.c:4675:gst_pad_send_event: Received event on flushing pad. Discarding 0:00:01.704047371 5174 0x1d712a0 INFO GST_EVENT gstpad.c:4675:gst_pad_send_event: Received event on flushing pad. Discarding 0:00:01.704058825 5174 0x1d712a0 INFO GST_EVENT gstpad.c:4675:gst_pad_send_event: Received event on flushing pad. Discarding 0:00:01.704073143 5174 0x1d712a0 WARN deinterleave deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() returned wrong-state 0:00:01.704371435 5174 0x1d712a0 WARN deinterleave deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() returned wrong-state 0:00:01.704564057 5174 0x1d712a0 WARN deinterleave deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() returned wrong-state 0:00:01.704730419 5174 0x1d712a0 WARN deinterleave deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() returned wrong-state 0:00:01.704894197 5174 0x1d712a0 WARN deinterleave deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() returned wrong-state 0:00:01.704937428 5174 0x1d712a0 INFO basesrc gstbasesrc.c:2278:gst_base_src_loop: pausing after gst_pad_push() = wrong-state What was the wrong in my setup? Could you please suggest how can I fix it to get the proper behavior (that new branch with appsink (fakesink) is added to pipeline when the new channel is recognized). Thank you very much in advance! Alexey Chernov From ensonic at hora-obscura.de Mon Sep 28 14:51:36 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Mon, 28 Sep 2009 15:51:36 +0300 Subject: [gst-devel] Problems with deinterleave plugin In-Reply-To: <200909281347.26913.4ernov@gmail.com> References: <200909281347.26913.4ernov@gmail.com> Message-ID: <4AC0B158.60500@hora-obscura.de> hi, I'll put some comments inline Alexey Chernov schrieb: > Hello, > > I'm working on a sound editor based on GStreamer and I faced the problem with > deinterleave plugin recently. > Is there project public? > To load and decode file I use pipeline: filesrc, decodebin, audioconvert, > audioresample, deinterleave and then several branches containing queue, > audioconvert and appsink (for the experiment I changed it to fakesink). > > Here's the code of main function: > > GMainLoop* _loop; > > GstElement* _pipeline; > > void load_file(const char* filename) > { > GstElement *source, *decodebin, *audio_convert, *audio_resample, *deint; > GstBus* bus; > > > > if (_loop) > { > g_main_loop_quit(_loop); > g_main_loop_unref(_loop); > } > _loop = g_main_loop_new (NULL, FALSE); > > /* Create gstreamer elements */ > if (_pipeline) > { > gst_element_set_state (_pipeline, GST_STATE_NULL); > gst_object_unref (GST_OBJECT (_pipeline)); > _pipeline=0; > } > _pipeline = gst_pipeline_new("decode_to_app"); > source = gst_element_factory_make("filesrc", "filesrc"); > decodebin = gst_element_factory_make("decodebin", "decode_bin"); > audio_convert = gst_element_factory_make("audioconvert","audio-convert"); > audio_resample = gst_element_factory_make("audioresample","audio-resample"); > deint = gst_element_factory_make("deinterleave", "deint"); > > > if (!_pipeline || !source || !decodebin || !audio_convert || !deint) > { > std::cerr<<"Elements could not be created. Exiting."< } > > /* Set up the pipeline */ > > /* we set the properties to the source element to receive only rtp packets*/ > g_object_set(G_OBJECT (source), "location", filename, NULL); > > /* we add a message handler */ > bus = gst_pipeline_get_bus (GST_PIPELINE (_pipeline)); > > gst_bus_add_watch (bus, bus_call, this); > gst_object_unref (bus); > > /* we add all elements into the pipeline */ > gst_bin_add_many (GST_BIN (_pipeline), source, decodebin, audio_convert, > audio_resample, deint, NULL); > > /* we link all the elements together */ > link_two_elements(source, decodebin); > link_two_elements(audio_resample, audio_convert); > link_two_elements(audio_convert,deint); > what is link_two_elements() doing differently from gst_element_link()? > > g_signal_connect (decodebin, "new-decoded-pad", G_CALLBACK (cb_new_pad), > audio_resample); > g_signal_connect (deint, "pad-added", G_CALLBACK (il_new_pad), 0); > > /* Set the pipeline to "playing" state*/ > gst_element_set_state (_pipeline, GST_STATE_PLAYING); > > /* Iterate */ > g_print ("Running...\n"); > g_main_loop_run (_loop); > > /* Out of the main loop, clean up nicely */ > g_print ("Returned, stopping listening\n"); > gst_element_set_state (_pipeline, GST_STATE_NULL); > > g_print ("Deleting pipeline\n"); > gst_object_unref (GST_OBJECT (_pipeline)); > } > > Here's il_new_pad implementation: > > > int _channels=0; > The global channels variable is a bit ugly (and might even be racy).. > void il_new_pad (GstElement *decodebin, GstPad *pad, gpointer data) > { > GstElement* element=0; > if (_pipeline) > { > GstElement *queue, *aconv, *ares, *appsink; > > char* num=itoa(_channels,num,10); > > char* name="queue"; > strcat(name,num); > queue = gst_element_factory_make("queue", name); > There is no need to do that. queue = gst_element_factory_make("queue", NULL); will make a unique name. > char* name="aconv"; > strcat(name,num); > aconv = gst_element_factory_make("audioconvert", name); > > char* name="sink"; > strcat(name,num); > appsink = gst_element_factory_make("fakesink", name); > > gst_bin_add_many (GST_BIN (_pipeline), queue, aconv, appsink, NULL); > Do gst_element_sync_state_with_parent() for each of the new elements. That hopefully fixes the warnings you see. Stefan > link_two_elements(queue, aconv); > link_two_elements(aconv,appsink); > > g_object_set(G_OBJECT (appsink), "sync", FALSE, NULL); > > element=queue; > > ++_channels; > } > > GstCaps *caps; > GstStructure *str; > GstPad *audiopad; > > /* only link once */ > audiopad = gst_element_get_static_pad (element, "sink"); > if (GST_PAD_IS_LINKED (audiopad)) > { > g_object_unref (audiopad); > } > > /* check media type */ > caps = gst_pad_get_caps (pad); > str = gst_caps_get_structure (caps, 0); > if (!g_strrstr (gst_structure_get_name (str), "audio")) > { > std::cerr<<"won't connect!"< gst_caps_unref (caps); > gst_object_unref (audiopad); > } > gst_caps_unref (caps); > > /* link'n'play */ > gst_pad_link (pad, audiopad); > } > > Everything seem to start OK, il_new_pad procedure works two times (for stereo > file), but then I've got the following messages in console: > > 0:00:01.703963841 5174 0x1d712a0 INFO GST_EVENT > gstpad.c:4675:gst_pad_send_event: Received event on flushing pad. > Discarding > > 0:00:01.703978717 5174 0x1d712a0 INFO GST_EVENT > gstpad.c:4675:gst_pad_send_event: Received event on flushing pad. > Discarding > > 0:00:01.703995479 5174 0x1d712a0 INFO GST_EVENT > gstpad.c:4675:gst_pad_send_event: Received event on flushing pad. > Discarding > > 0:00:01.704007213 5174 0x1d712a0 INFO GST_EVENT > gstpad.c:4675:gst_pad_send_event: Received event on flushing pad. > Discarding > > 0:00:01.704021111 5174 0x1d712a0 INFO GST_EVENT > gstpad.c:4675:gst_pad_send_event: Received event on flushing pad. > Discarding > > 0:00:01.704032565 5174 0x1d712a0 INFO GST_EVENT > gstpad.c:4675:gst_pad_send_event: Received event on flushing pad. > Discarding > > 0:00:01.704047371 5174 0x1d712a0 INFO GST_EVENT > gstpad.c:4675:gst_pad_send_event: Received event on flushing pad. > Discarding > > 0:00:01.704058825 5174 0x1d712a0 INFO GST_EVENT > gstpad.c:4675:gst_pad_send_event: Received event on flushing pad. > Discarding > > 0:00:01.704073143 5174 0x1d712a0 WARN deinterleave > deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() returned > wrong-state > > 0:00:01.704371435 5174 0x1d712a0 WARN deinterleave > deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() returned > wrong-state > > 0:00:01.704564057 5174 0x1d712a0 WARN deinterleave > deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() returned > wrong-state > > 0:00:01.704730419 5174 0x1d712a0 WARN deinterleave > deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() returned > wrong-state > > 0:00:01.704894197 5174 0x1d712a0 WARN deinterleave > deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() returned > wrong-state > > 0:00:01.704937428 5174 0x1d712a0 INFO basesrc > gstbasesrc.c:2278:gst_base_src_loop: pausing after gst_pad_push() = > wrong-state > > What was the wrong in my setup? Could you please suggest how can I fix it to > get the proper behavior (that new branch with appsink (fakesink) is added to > pipeline when the new channel is recognized). > > Thank you very much in advance! > > Alexey Chernov > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From ensonic at hora-obscura.de Mon Sep 28 14:40:57 2009 From: ensonic at hora-obscura.de (Stefan Kost) Date: Mon, 28 Sep 2009 15:40:57 +0300 Subject: [gst-devel] unable to load pipeline from XML In-Reply-To: <6c9581cb0909251137v15601b0m5bfede30a2f181a@mail.gmail.com> References: <6c9581cb0909251137v15601b0m5bfede30a2f181a@mail.gmail.com> Message-ID: <4AC0AED9.6030002@hora-obscura.de> hi, Ottaviano Vasselli schrieb: > Hello, > I'm having trouble with gst-xmllaunch. When I launch that simple > pipeline with gst-launch, everything goes fine: > > gst-launch-0.10 filesrc location=$SRC_FILE ! decodebin name=decode decode. \ > ! queue ! theoraenc ! oggmux name=mux decode. ! queue ! > audioconvert ! vorbisenc \ > ! mux. mux. ! filesink location=$DST_FILE > > (with $SRC_FILE and $DST_FILE existing file names) > > so I added "-o saved_pipe.xml" obtaining an xml file looking valid. > But when I run gst-xmllaunch -vvv saved_pipe.xml , transcoding does > not start and pipeline terminates with following output: > xml serialisation has some limmitations and is not widley used. In the above case it might fail to handle the dynamic pads. You could use G_DEBUG=fatal_warnings gdb --args gst-xmllaunch -vvv saved_pipe.xml and break on the error. Get the backtrace, figure a fix, report problem+backtrace+patch to bugzilla. Stefan > (gst-launch-0.10:3746): GStreamer-CRITICAL **: pad sink_563092733 > is not a source pad > > (gst-launch-0.10:3746): GStreamer-WARNING **: Name fakesink is not > unique in bin decode, not adding > > (gst-launch-0.10:3746): GStreamer-WARNING **: Name typefind is not > unique in bin decode, not adding > francesco at helium:/tmp$ gst-xmllaunch-0.10 -vvv test.xml > > (gst-launch-0.10:3758): GStreamer-CRITICAL **: pad sink_563092733 > is not a source pad > > (gst-launch-0.10:3758): GStreamer-WARNING **: Name fakesink is not > unique in bin decode, not adding > > (gst-launch-0.10:3758): GStreamer-WARNING **: Name typefind is not > unique in bin decode, not adding > > (gst-launch-0.10:3758): GStreamer-CRITICAL **: > gst_pad_load_and_link: assertion `pad != NULL' failed > Setting pipeline to PAUSED ... > /pipeline0/decode/typefind.src: caps = video/x-msvideo > Pipeline is PREROLLING ... > ... > > > Also tried adding filesrc0.location="..." filesink0.location="..." > with real files specified, but no change. > Does somebody have any clue about? > > > Thank you for your attention, bye > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From daltoncezane at gmail.com Mon Sep 28 17:32:58 2009 From: daltoncezane at gmail.com (=?ISO-8859-1?Q?Dalton_C=E9zane?=) Date: Mon, 28 Sep 2009 12:32:58 -0300 Subject: [gst-devel] Demux H264 video and HE-AAC audio (Edward Hervey) In-Reply-To: <98029fd50909251449w6d13ee21kab5657baf83aab00@mail.gmail.com> References: <98029fd50909251449w6d13ee21kab5657baf83aab00@mail.gmail.com> Message-ID: <98029fd50909280832v76256622w3a886996a2c194cf@mail.gmail.com> No one? Maybe the problem is with the he-aac decoder "faad"... does anyone suggest me another? I did not find... :( Thanks in advance! On Fri, Sep 25, 2009 at 6:49 PM, Dalton C?zane wrote: > I tried to run h264 video and he-aac audio in another machine with > mpegtsdemux, but it did not work: > > /home/dalton# gst-launch-0.10 filesrc location="GLOBO188bytes.mpg" ! > mpegtsdemux name=demux demux. queue ! ffdec_h264 ! xvimagesink demux. queue > ! faad ! audioconvert ! alsasink > Setting pipeline to PAUSED ... > Pipeline is PREROLLING ... > > ..... > And continues like it was in loop... > > Can anyone help me? > > > _____________________________________________________________________________________________________ > > Use mpegtsdemux and not ffdemux_mpegts. The rule of thumb is : don't use > the ffmpeg demuxers (they have a rank of NONE for a reason). > > Edward > > > On Fri, 2009-09-18 at 16:42 -0300, Dalton C?zane wrote: > > Hi all, > > I am new at list and at the GStreamer study. I am trying to demux H264 > > video and HE-AAC audio with gst-launch-0.10. > > I already succeeded just the video, without audio, with this command > > line: gst-launch-0.10 filesrc location="GLOBO188bytes.mpg" ! > > ffdemux_mpegts name=demux demux. queue ! ffdec_h264 ! xvimagesink ! > > demux. queue ! faad ! audioconvert ! osssink > > > > This way, the video is displayed but the sound does not play. > > Can anyone help me? Some tip? > > > > Thanks in advance. > > -- > ======================================================= > Dalton C?zane - Voip UFCG: 1075-2005 > Mestrando em Ci?ncia da Computa??o (UFCG) > Bacharel em Ci?ncia da Computa??o (UFCG) > T?cnico em Inform?tica (ETER) > -- ======================================================= Dalton C?zane - Voip UFCG: 1075-2005 Mestrando em Ci?ncia da Computa??o (UFCG) Bacharel em Ci?ncia da Computa??o (UFCG) T?cnico em Inform?tica (ETER) -------------- next part -------------- An HTML attachment was scrubbed... URL: From 4ernov at gmail.com Mon Sep 28 18:34:19 2009 From: 4ernov at gmail.com (Alexey Chernov) Date: Mon, 28 Sep 2009 20:34:19 +0400 Subject: [gst-devel] Problems with deinterleave plugin In-Reply-To: <4AC0B158.60500@hora-obscura.de> References: <200909281347.26913.4ernov@gmail.com> <4AC0B158.60500@hora-obscura.de> Message-ID: <200909282034.19500.4ernov@gmail.com> Hi Stefan, Thank you very much for help, I'll try your suggestions. The project is targeted to be public and OSS, of course (it's Laudi, laudi.sourceforge.net) but I'm afraid not mature enough for any releases yet.. There're some ugly places in my code it's because the project itself is on C++ but I removed any class stuff and make it like global variables. link_two_elements() is this: void link_two_elements(GstElement* src_element, GstElement* sink_element) { if(!gst_element_link(src_element, sink_element)) g_debug("Error linking %s to %s", gst_element_get_name(src_element), gst_element_get_name(sink_element)); } I also wanted to ask: can these warnings I wrote lead to hang up of the pipeline? After them it's like it is paused, no messages and no progress, but in GST_LEVEL_DEBUG level the output is as it plays the same chunk of file (as of timestamps). Thanks again for support, it was the big problem for me.. ? ????????? ?? ??????????? 28 ???????? 2009 16:51:36 ????? Stefan Kost ???????: > hi, > > I'll put some comments inline > > Alexey Chernov schrieb: > > Hello, > > > > I'm working on a sound editor based on GStreamer and I faced the problem > > with deinterleave plugin recently. > > Is there project public? > > > To load and decode file I use pipeline: filesrc, decodebin, audioconvert, > > audioresample, deinterleave and then several branches containing queue, > > audioconvert and appsink (for the experiment I changed it to fakesink). > > > > Here's the code of main function: > > > > GMainLoop* _loop; > > > > GstElement* _pipeline; > > > > void load_file(const char* filename) > > { > > GstElement *source, *decodebin, *audio_convert, *audio_resample, *deint; > > GstBus* bus; > > > > > > > > if (_loop) > > { > > g_main_loop_quit(_loop); > > g_main_loop_unref(_loop); > > } > > _loop = g_main_loop_new (NULL, FALSE); > > > > /* Create gstreamer elements */ > > if (_pipeline) > > { > > gst_element_set_state (_pipeline, GST_STATE_NULL); > > gst_object_unref (GST_OBJECT (_pipeline)); > > _pipeline=0; > > } > > _pipeline = gst_pipeline_new("decode_to_app"); > > source = gst_element_factory_make("filesrc", "filesrc"); > > decodebin = gst_element_factory_make("decodebin", "decode_bin"); > > audio_convert = > > gst_element_factory_make("audioconvert","audio-convert"); audio_resample > > = gst_element_factory_make("audioresample","audio-resample"); deint = > > gst_element_factory_make("deinterleave", "deint"); > > > > > > if (!_pipeline || !source || !decodebin || !audio_convert || !deint) > > { > > std::cerr<<"Elements could not be created. Exiting."< > } > > > > /* Set up the pipeline */ > > > > /* we set the properties to the source element to receive only rtp > > packets*/ g_object_set(G_OBJECT (source), "location", filename, NULL); > > > > /* we add a message handler */ > > bus = gst_pipeline_get_bus (GST_PIPELINE (_pipeline)); > > > > gst_bus_add_watch (bus, bus_call, this); > > gst_object_unref (bus); > > > > /* we add all elements into the pipeline */ > > gst_bin_add_many (GST_BIN (_pipeline), source, decodebin, audio_convert, > > audio_resample, deint, NULL); > > > > /* we link all the elements together */ > > link_two_elements(source, decodebin); > > link_two_elements(audio_resample, audio_convert); > > link_two_elements(audio_convert,deint); > > what is link_two_elements() doing differently from gst_element_link()? > > > g_signal_connect (decodebin, "new-decoded-pad", G_CALLBACK (cb_new_pad), > > audio_resample); > > g_signal_connect (deint, "pad-added", G_CALLBACK (il_new_pad), 0); > > > > /* Set the pipeline to "playing" state*/ > > gst_element_set_state (_pipeline, GST_STATE_PLAYING); > > > > /* Iterate */ > > g_print ("Running...\n"); > > g_main_loop_run (_loop); > > > > /* Out of the main loop, clean up nicely */ > > g_print ("Returned, stopping listening\n"); > > gst_element_set_state (_pipeline, GST_STATE_NULL); > > > > g_print ("Deleting pipeline\n"); > > gst_object_unref (GST_OBJECT (_pipeline)); > > } > > > > Here's il_new_pad implementation: > > > > > > int _channels=0; > > The global channels variable is a bit ugly (and might even be racy).. > > > void il_new_pad (GstElement *decodebin, GstPad *pad, gpointer data) > > { > > GstElement* element=0; > > if (_pipeline) > > { > > GstElement *queue, *aconv, *ares, *appsink; > > > > char* num=itoa(_channels,num,10); > > > > char* name="queue"; > > strcat(name,num); > > queue = gst_element_factory_make("queue", name); > > There is no need to do that. > queue = gst_element_factory_make("queue", NULL); > will make a unique name. > > > char* name="aconv"; > > strcat(name,num); > > aconv = gst_element_factory_make("audioconvert", name); > > > > char* name="sink"; > > strcat(name,num); > > appsink = gst_element_factory_make("fakesink", name); > > > > gst_bin_add_many (GST_BIN (_pipeline), queue, aconv, appsink, NULL); > > Do gst_element_sync_state_with_parent() for each of the new elements. > That hopefully fixes the warnings you see. > > Stefan > > > link_two_elements(queue, aconv); > > link_two_elements(aconv,appsink); > > > > g_object_set(G_OBJECT (appsink), "sync", FALSE, NULL); > > > > element=queue; > > > > ++_channels; > > } > > > > GstCaps *caps; > > GstStructure *str; > > GstPad *audiopad; > > > > /* only link once */ > > audiopad = gst_element_get_static_pad (element, "sink"); > > if (GST_PAD_IS_LINKED (audiopad)) > > { > > g_object_unref (audiopad); > > } > > > > /* check media type */ > > caps = gst_pad_get_caps (pad); > > str = gst_caps_get_structure (caps, 0); > > if (!g_strrstr (gst_structure_get_name (str), "audio")) > > { > > std::cerr<<"won't connect!"< > gst_caps_unref (caps); > > gst_object_unref (audiopad); > > } > > gst_caps_unref (caps); > > > > /* link'n'play */ > > gst_pad_link (pad, audiopad); > > } > > > > Everything seem to start OK, il_new_pad procedure works two times (for > > stereo file), but then I've got the following messages in console: > > > > 0:00:01.703963841 5174 0x1d712a0 INFO GST_EVENT > > gstpad.c:4675:gst_pad_send_event: Received event on flushing > > pad. Discarding > > > > 0:00:01.703978717 5174 0x1d712a0 INFO GST_EVENT > > gstpad.c:4675:gst_pad_send_event: Received event on flushing > > pad. Discarding > > > > 0:00:01.703995479 5174 0x1d712a0 INFO GST_EVENT > > gstpad.c:4675:gst_pad_send_event: Received event on flushing > > pad. Discarding > > > > 0:00:01.704007213 5174 0x1d712a0 INFO GST_EVENT > > gstpad.c:4675:gst_pad_send_event: Received event on flushing > > pad. Discarding > > > > 0:00:01.704021111 5174 0x1d712a0 INFO GST_EVENT > > gstpad.c:4675:gst_pad_send_event: Received event on flushing > > pad. Discarding > > > > 0:00:01.704032565 5174 0x1d712a0 INFO GST_EVENT > > gstpad.c:4675:gst_pad_send_event: Received event on flushing > > pad. Discarding > > > > 0:00:01.704047371 5174 0x1d712a0 INFO GST_EVENT > > gstpad.c:4675:gst_pad_send_event: Received event on flushing > > pad. Discarding > > > > 0:00:01.704058825 5174 0x1d712a0 INFO GST_EVENT > > gstpad.c:4675:gst_pad_send_event: Received event on flushing > > pad. Discarding > > > > 0:00:01.704073143 5174 0x1d712a0 WARN deinterleave > > deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() > > returned wrong-state > > > > 0:00:01.704371435 5174 0x1d712a0 WARN deinterleave > > deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() > > returned wrong-state > > > > 0:00:01.704564057 5174 0x1d712a0 WARN deinterleave > > deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() > > returned wrong-state > > > > 0:00:01.704730419 5174 0x1d712a0 WARN deinterleave > > deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() > > returned wrong-state > > > > 0:00:01.704894197 5174 0x1d712a0 WARN deinterleave > > deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() > > returned wrong-state > > > > 0:00:01.704937428 5174 0x1d712a0 INFO basesrc > > gstbasesrc.c:2278:gst_base_src_loop: pausing after > > gst_pad_push() = wrong-state > > > > What was the wrong in my setup? Could you please suggest how can I fix it > > to get the proper behavior (that new branch with appsink (fakesink) is > > added to pipeline when the new channel is recognized). > > > > Thank you very much in advance! > > > > Alexey Chernov > > > > ------------------------------------------------------------------------- > >----- Come build with us! The BlackBerry® Developer Conference in SF, > > CA is the only developer event you need to attend this year. Jumpstart > > your developing skills, take BlackBerry mobile applications to market and > > stay ahead of the curve. Join us from November 9-12, 2009. Register > > now! http://p.sf.net/sfu/devconf > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From startoftext at gmail.com Mon Sep 28 20:04:25 2009 From: startoftext at gmail.com (James Pearson) Date: Mon, 28 Sep 2009 13:04:25 -0500 Subject: [gst-devel] adding tee to alsasink causing problems Message-ID: <9830AAEA-6066-4E4D-AA3E-EFFF3B9360FF@gmail.com> So here is my pipeline. The point is to record audio and video and display it at the same time. v4l2src -> tee -> video/x-raw-yuv,width=320,height=240 -> ffmpegcolorspace -> ffenc_flv -> ffmux_flv -> filesink -> xvimagesink alsasrc -> tee > audioresample -> audio-x-raw-int,rate=44100 - -> audioconvert -> lame -> ffmux_flv (same mux as above) -> alsasink I have been using this pipeline minus the tee for live playback (alsasink,xvimagesink) for some time and it is working well. The video part, even with the live display works ok but when i add in the tee and alsasink the pipeline gets stuck with nothing going to disk and the video displayed is stuck. Even just adding the alsasink to the pipeline without connecting it causes this result. Any suggestions? I already tried adding queues to almost everything. Thanks -James- From 4ernov at gmail.com Tue Sep 29 00:06:38 2009 From: 4ernov at gmail.com (Alexey Chernov) Date: Tue, 29 Sep 2009 02:06:38 +0400 Subject: [gst-devel] Problems with deinterleave plugin In-Reply-To: <4AC0B158.60500@hora-obscura.de> References: <200909281347.26913.4ernov@gmail.com> <4AC0B158.60500@hora-obscura.de> Message-ID: <200909290206.38582.4ernov@gmail.com> Thank you very much! I've added gst_element_sync_state_with_parent() and now it works perfect without such warnings. Thank you, Stefan! ? ????????? ?? ??????????? 28 ???????? 2009 16:51:36 ????? Stefan Kost ???????: > hi, > > I'll put some comments inline > > Alexey Chernov schrieb: > > Hello, > > > > I'm working on a sound editor based on GStreamer and I faced the problem > > with deinterleave plugin recently. > > Is there project public? > > > To load and decode file I use pipeline: filesrc, decodebin, audioconvert, > > audioresample, deinterleave and then several branches containing queue, > > audioconvert and appsink (for the experiment I changed it to fakesink). > > > > Here's the code of main function: > > > > GMainLoop* _loop; > > > > GstElement* _pipeline; > > > > void load_file(const char* filename) > > { > > GstElement *source, *decodebin, *audio_convert, *audio_resample, *deint; > > GstBus* bus; > > > > > > > > if (_loop) > > { > > g_main_loop_quit(_loop); > > g_main_loop_unref(_loop); > > } > > _loop = g_main_loop_new (NULL, FALSE); > > > > /* Create gstreamer elements */ > > if (_pipeline) > > { > > gst_element_set_state (_pipeline, GST_STATE_NULL); > > gst_object_unref (GST_OBJECT (_pipeline)); > > _pipeline=0; > > } > > _pipeline = gst_pipeline_new("decode_to_app"); > > source = gst_element_factory_make("filesrc", "filesrc"); > > decodebin = gst_element_factory_make("decodebin", "decode_bin"); > > audio_convert = > > gst_element_factory_make("audioconvert","audio-convert"); audio_resample > > = gst_element_factory_make("audioresample","audio-resample"); deint = > > gst_element_factory_make("deinterleave", "deint"); > > > > > > if (!_pipeline || !source || !decodebin || !audio_convert || !deint) > > { > > std::cerr<<"Elements could not be created. Exiting."< > } > > > > /* Set up the pipeline */ > > > > /* we set the properties to the source element to receive only rtp > > packets*/ g_object_set(G_OBJECT (source), "location", filename, NULL); > > > > /* we add a message handler */ > > bus = gst_pipeline_get_bus (GST_PIPELINE (_pipeline)); > > > > gst_bus_add_watch (bus, bus_call, this); > > gst_object_unref (bus); > > > > /* we add all elements into the pipeline */ > > gst_bin_add_many (GST_BIN (_pipeline), source, decodebin, audio_convert, > > audio_resample, deint, NULL); > > > > /* we link all the elements together */ > > link_two_elements(source, decodebin); > > link_two_elements(audio_resample, audio_convert); > > link_two_elements(audio_convert,deint); > > what is link_two_elements() doing differently from gst_element_link()? > > > g_signal_connect (decodebin, "new-decoded-pad", G_CALLBACK (cb_new_pad), > > audio_resample); > > g_signal_connect (deint, "pad-added", G_CALLBACK (il_new_pad), 0); > > > > /* Set the pipeline to "playing" state*/ > > gst_element_set_state (_pipeline, GST_STATE_PLAYING); > > > > /* Iterate */ > > g_print ("Running...\n"); > > g_main_loop_run (_loop); > > > > /* Out of the main loop, clean up nicely */ > > g_print ("Returned, stopping listening\n"); > > gst_element_set_state (_pipeline, GST_STATE_NULL); > > > > g_print ("Deleting pipeline\n"); > > gst_object_unref (GST_OBJECT (_pipeline)); > > } > > > > Here's il_new_pad implementation: > > > > > > int _channels=0; > > The global channels variable is a bit ugly (and might even be racy).. > > > void il_new_pad (GstElement *decodebin, GstPad *pad, gpointer data) > > { > > GstElement* element=0; > > if (_pipeline) > > { > > GstElement *queue, *aconv, *ares, *appsink; > > > > char* num=itoa(_channels,num,10); > > > > char* name="queue"; > > strcat(name,num); > > queue = gst_element_factory_make("queue", name); > > There is no need to do that. > queue = gst_element_factory_make("queue", NULL); > will make a unique name. > > > char* name="aconv"; > > strcat(name,num); > > aconv = gst_element_factory_make("audioconvert", name); > > > > char* name="sink"; > > strcat(name,num); > > appsink = gst_element_factory_make("fakesink", name); > > > > gst_bin_add_many (GST_BIN (_pipeline), queue, aconv, appsink, NULL); > > Do gst_element_sync_state_with_parent() for each of the new elements. > That hopefully fixes the warnings you see. > > Stefan > > > link_two_elements(queue, aconv); > > link_two_elements(aconv,appsink); > > > > g_object_set(G_OBJECT (appsink), "sync", FALSE, NULL); > > > > element=queue; > > > > ++_channels; > > } > > > > GstCaps *caps; > > GstStructure *str; > > GstPad *audiopad; > > > > /* only link once */ > > audiopad = gst_element_get_static_pad (element, "sink"); > > if (GST_PAD_IS_LINKED (audiopad)) > > { > > g_object_unref (audiopad); > > } > > > > /* check media type */ > > caps = gst_pad_get_caps (pad); > > str = gst_caps_get_structure (caps, 0); > > if (!g_strrstr (gst_structure_get_name (str), "audio")) > > { > > std::cerr<<"won't connect!"< > gst_caps_unref (caps); > > gst_object_unref (audiopad); > > } > > gst_caps_unref (caps); > > > > /* link'n'play */ > > gst_pad_link (pad, audiopad); > > } > > > > Everything seem to start OK, il_new_pad procedure works two times (for > > stereo file), but then I've got the following messages in console: > > > > 0:00:01.703963841 5174 0x1d712a0 INFO GST_EVENT > > gstpad.c:4675:gst_pad_send_event: Received event on flushing > > pad. Discarding > > > > 0:00:01.703978717 5174 0x1d712a0 INFO GST_EVENT > > gstpad.c:4675:gst_pad_send_event: Received event on flushing > > pad. Discarding > > > > 0:00:01.703995479 5174 0x1d712a0 INFO GST_EVENT > > gstpad.c:4675:gst_pad_send_event: Received event on flushing > > pad. Discarding > > > > 0:00:01.704007213 5174 0x1d712a0 INFO GST_EVENT > > gstpad.c:4675:gst_pad_send_event: Received event on flushing > > pad. Discarding > > > > 0:00:01.704021111 5174 0x1d712a0 INFO GST_EVENT > > gstpad.c:4675:gst_pad_send_event: Received event on flushing > > pad. Discarding > > > > 0:00:01.704032565 5174 0x1d712a0 INFO GST_EVENT > > gstpad.c:4675:gst_pad_send_event: Received event on flushing > > pad. Discarding > > > > 0:00:01.704047371 5174 0x1d712a0 INFO GST_EVENT > > gstpad.c:4675:gst_pad_send_event: Received event on flushing > > pad. Discarding > > > > 0:00:01.704058825 5174 0x1d712a0 INFO GST_EVENT > > gstpad.c:4675:gst_pad_send_event: Received event on flushing > > pad. Discarding > > > > 0:00:01.704073143 5174 0x1d712a0 WARN deinterleave > > deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() > > returned wrong-state > > > > 0:00:01.704371435 5174 0x1d712a0 WARN deinterleave > > deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() > > returned wrong-state > > > > 0:00:01.704564057 5174 0x1d712a0 WARN deinterleave > > deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() > > returned wrong-state > > > > 0:00:01.704730419 5174 0x1d712a0 WARN deinterleave > > deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() > > returned wrong-state > > > > 0:00:01.704894197 5174 0x1d712a0 WARN deinterleave > > deinterleave.c:810:gst_deinterleave_process: gst_pad_alloc_buffer() > > returned wrong-state > > > > 0:00:01.704937428 5174 0x1d712a0 INFO basesrc > > gstbasesrc.c:2278:gst_base_src_loop: pausing after > > gst_pad_push() = wrong-state > > > > What was the wrong in my setup? Could you please suggest how can I fix it > > to get the proper behavior (that new branch with appsink (fakesink) is > > added to pipeline when the new channel is recognized). > > > > Thank you very much in advance! > > > > Alexey Chernov > > > > ------------------------------------------------------------------------- > >----- Come build with us! The BlackBerry® Developer Conference in SF, > > CA is the only developer event you need to attend this year. Jumpstart > > your developing skills, take BlackBerry mobile applications to market and > > stay ahead of the curve. Join us from November 9-12, 2009. Register > > now! http://p.sf.net/sfu/devconf > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From dxssx.dxssx at gmail.com Tue Sep 29 05:36:21 2009 From: dxssx.dxssx at gmail.com (dxssx) Date: Tue, 29 Sep 2009 11:36:21 +0800 Subject: [gst-devel] qtdemux bug? Message-ID: Hi, I have got some problems when using qtdemux. Pipeline "filesrc location=xxx.mp4 ! qtdemux name=demux demux.audio_00 ! queue ! filesink location=a.aac demux.video_00 ! queue ! filesink location=a.h264" works well, but "filesrc location=xxx.mp4 ! qtdemux name=demux demux.audio_00 ! filesink location=a.aac demux.video_00 ! filesink location=a.h264" does not work. It seems to be a dead lock in basesink which is waiting for a play event. Does it necessary to add a queue in pipeline when demuxing? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: From sledgehammer_999 at hotmail.com Tue Sep 29 07:34:15 2009 From: sledgehammer_999 at hotmail.com (sledge hammer) Date: Tue, 29 Sep 2009 08:34:15 +0300 Subject: [gst-devel] Demux H264 video and HE-AAC audio (Edward Hervey) In-Reply-To: <98029fd50909280832v76256622w3a886996a2c194cf@mail.gmail.com> References: <98029fd50909251449w6d13ee21kab5657baf83aab00@mail.gmail.com> Message-ID: Does the file playback correctly if you use the playbin element? If not, then you should file a bug and attach your video file. gst-launch-0.10 playbin uri=file:///"path to file" (the 3rd '/' is for the root directory) Date: Mon, 28 Sep 2009 12:32:58 -0300 From: daltoncezane at gmail.com To: gstreamer-devel at lists.sourceforge.net Subject: Re: [gst-devel] Demux H264 video and HE-AAC audio (Edward Hervey) No one? Maybe the problem is with the he-aac decoder "faad"... does anyone suggest me another? I did not find... :( Thanks in advance! On Fri, Sep 25, 2009 at 6:49 PM, Dalton C?zane wrote: I tried to run h264 video and he-aac audio in another machine with mpegtsdemux, but it did not work: /home/dalton# gst-launch-0.10 filesrc location="GLOBO188bytes.mpg" ! mpegtsdemux name=demux demux. queue ! ffdec_h264 ! xvimagesink demux. queue ! faad ! audioconvert ! alsasink Setting pipeline to PAUSED ... Pipeline is PREROLLING ... ..... And continues like it was in loop... Can anyone help me? _____________________________________________________________________________________________________ Use mpegtsdemux and not ffdemux_mpegts. The rule of thumb is : don't use the ffmpeg demuxers (they have a rank of NONE for a reason). Edward On Fri, 2009-09-18 at 16:42 -0300, Dalton C?zane wrote: > Hi all, > I am new at list and at the GStreamer study. I am trying to demux H264 > video and HE-AAC audio with gst-launch-0.10. > I already succeeded just the video, without audio, with this command > line: gst-launch-0.10 filesrc location="GLOBO188bytes.mpg" ! > ffdemux_mpegts name=demux demux. queue ! ffdec_h264 ! xvimagesink ! > demux. queue ! faad ! audioconvert ! osssink > > This way, the video is displayed but the sound does not play. > Can anyone help me? Some tip? > > Thanks in advance. -- ======================================================= Dalton C?zane - Voip UFCG: 1075-2005 Mestrando em Ci?ncia da Computa??o (UFCG) Bacharel em Ci?ncia da Computa??o (UFCG) T?cnico em Inform?tica (ETER) -- ======================================================= Dalton C?zane - Voip UFCG: 1075-2005 Mestrando em Ci?ncia da Computa??o (UFCG) Bacharel em Ci?ncia da Computa??o (UFCG) T?cnico em Inform?tica (ETER) _________________________________________________________________ ?? What?s New ??? ????????? ????? ??? ???? ?????????. ?????? ???. http://home.live.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From t.i.m at zen.co.uk Tue Sep 29 10:21:28 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Tue, 29 Sep 2009 09:21:28 +0100 Subject: [gst-devel] qtdemux bug? In-Reply-To: References: Message-ID: <1254212488.5699.1.camel@zingle> On Tue, 2009-09-29 at 11:36 +0800, dxssx wrote: > Pipeline "filesrc location=xxx.mp4 ! qtdemux name=demux > demux.audio_00 ! queue ! filesink location=a.aac demux.video_00 ! > queue ! filesink location=a.h264" works well, but "filesrc > location=xxx.mp4 ! qtdemux name=demux demux.audio_00 ! filesink > location=a.aac demux.video_00 ! filesink location=a.h264" does not > work. > > > It seems to be a dead lock in basesink which is waiting for a play > event. > > > Does it necessary to add a queue in pipeline when demuxing? Yes, you need to add queues, otherwise the pipeline can't preroll because the first sink to get data will block and control won't be given back to qtdemux to push data to the other sink (which will then wait for data forever and never preroll). Cheers -Tim > From dirk.griffioen at barcelonamedia.org Tue Sep 29 11:23:53 2009 From: dirk.griffioen at barcelonamedia.org (Dirk Griffioen) Date: Tue, 29 Sep 2009 11:23:53 +0200 Subject: [gst-devel] frei0r examples Message-ID: <4AC1D229.9000108@barcelonamedia.org> Hi, I was wondering if there are any frei0r examples, for instance on how to use the delay. Thanks in advance. Best, Dirk From sebastian.droege at collabora.co.uk Tue Sep 29 11:35:29 2009 From: sebastian.droege at collabora.co.uk (Sebastian =?ISO-8859-1?Q?Dr=F6ge?=) Date: Tue, 29 Sep 2009 11:35:29 +0200 Subject: [gst-devel] frei0r examples In-Reply-To: <4AC1D229.9000108@barcelonamedia.org> References: <4AC1D229.9000108@barcelonamedia.org> Message-ID: <1254216930.4853.10.camel@odin.lan> Am Dienstag, den 29.09.2009, 11:23 +0200 schrieb Dirk Griffioen: > Hi, > > I was wondering if there are any frei0r examples, for instance on how to > use the delay. There are no real examples yet... for using the delay0r plugin: gst-launch-0.10 videotestsrc ! ffmpegcolorspace ! frei0r-filter-delay0r delaytime=10 ! ffmpegcolorspace ! xvimagesink That will delay the video for 10 seconds. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Dies ist ein digital signierter Nachrichtenteil URL: From rohan at perzonae.com Tue Sep 29 12:04:01 2009 From: rohan at perzonae.com (Rohan) Date: Tue, 29 Sep 2009 12:04:01 +0200 Subject: [gst-devel] works with gst-launch but not in python Message-ID: <4AC1DB91.9050400@perzonae.com> Hi all, I am trying to stream video and audio over the network. I have the video working so far thanks to pointing out my little blunder the other day. But audio is not working, and this time I have remembered to add the src to the pipeline. The gst-launch sequence works perfectly, so I am obviously missing another obvious thing. These are the gst-launch commands that play happily together: sender: gst-launch-010 audiotestsrc ! queue ! audioconvert ! speexenc ! \ tcpserversink host=127.0.0.1 port=5001 receiver: gst-launch-0.10 tcpclientsrc host=127.0.0.1 port=5001 ! speexdec ! \ queue ! alsasink sync=false And here are the python scripts that are just reiterating this in python, but it is not working ------------------------------------------------------------------- #!/bin/env python # sound_receiver.py #gst-launch-0.10 tcpclientsrc host=127.0.0.1 port=5001 ! speexdec ! queue ! alsasink sync=false import gobject import pygst pygst.require("0.10") import gst class server(object): def __init__(self): pipe = gst.Pipeline("receive") src = gst.element_factory_make("tcpclientsrc") src.set_property("host", '127.0.0.1') src.set_property("port", 5001) pipe.add(src) sbin = self.buildsound() pipe.add(sbin) src.link(sbin) pipe.set_state(gst.STATE_PLAYING ) def buildsound(self): bin = gst.Bin("sound") speexdec = gst.element_factory_make("speexdec") queue = gst.element_factory_make('queue') soundsink = gst.element_factory_make("autoaudiosink") bin.add(speexdec, queue, soundsink) gst.element_link_many(speexdec, queue, soundsink) binsink = gst.GhostPad("sbinsink", speexdec.get_pad("sink")) bin.add_pad(binsink) return bin if __name__ == "__main__": server() loop = gobject.MainLoop() loop.run() ---------------------------------------------------------------------- #!/bin/env python # sound_sender.py # gst-launch-010 audiotestsrc ! queue ! audioconvert ! speexenc ! tcpserversink host=127.0.0.1 port=5001 import gobject import pygst pygst.require("0.10") import gst class client(object): def __init__(self): pipe = gst.Pipeline("sound") src = gst.element_factory_make("audiotestsrc") sbin = self.build_sound() pipe.add(src, sbin) src.link(sbin) pipe.set_state(gst.STATE_PLAYING) def build_sound(self): bin = gst.Bin("sbin") queue = gst.element_factory_make("queue") decode = gst.element_factory_make("decodebin") convert = gst.element_factory_make("audioconvert") speexenc = gst.element_factory_make("speexenc") sink = gst.element_factory_make("tcpserversink") sink.set_property("host", '127.0.0.1') sink.set_property("port", 5001) bin.add(queue, decode, convert, speexenc, sink) gst.element_link_many(queue, convert, speexenc, sink) binsink = gst.GhostPad("sbinsink", queue.get_pad("sink")) bin.add_pad(binsink) return bin if __name__ == '__main__': client() loop = gobject.MainLoop() loop.run() -------------------------------------------------------------------- I have noticed that with the gst-launch sequences you must have the sound sender running before starting the sound receiver, but starting the python scripts in either order makes no difference to the outcome: no sound, and no error output. Thanks in advance for any help, this has had me stumped for a while now, Rohan From dirk.griffioen at barcelonamedia.org Tue Sep 29 12:06:48 2009 From: dirk.griffioen at barcelonamedia.org (Dirk Griffioen) Date: Tue, 29 Sep 2009 12:06:48 +0200 Subject: [gst-devel] frei0r examples In-Reply-To: <1254216930.4853.10.camel@odin.lan> References: <4AC1D229.9000108@barcelonamedia.org> <1254216930.4853.10.camel@odin.lan> Message-ID: <4AC1DC38.8090005@barcelonamedia.org> Thanks Sebastian! I just found alomost the same line :) But your second ffmpegcolorspace really does the trick ... Cheers! > Am Dienstag, den 29.09.2009, 11:23 +0200 schrieb Dirk Griffioen: > >> Hi, >> >> I was wondering if there are any frei0r examples, for instance on how to >> use the delay. >> > > There are no real examples yet... for using the delay0r plugin: > > gst-launch-0.10 videotestsrc ! ffmpegcolorspace ! frei0r-filter-delay0r > delaytime=10 ! ffmpegcolorspace ! xvimagesink > > That will delay the video for 10 seconds. > From rohan at perzonae.com Tue Sep 29 12:55:30 2009 From: rohan at perzonae.com (Rohan) Date: Tue, 29 Sep 2009 12:55:30 +0200 Subject: [gst-devel] adding tee to alsasink causing problems In-Reply-To: <9830AAEA-6066-4E4D-AA3E-EFFF3B9360FF@gmail.com> References: <9830AAEA-6066-4E4D-AA3E-EFFF3B9360FF@gmail.com> Message-ID: <4AC1E7A2.9050806@perzonae.com> Hi James, I am no expert but could you put the actual gst-launch commands you are using, because it is not clear here where you use the tee twice, for either. I have been meddling with splitting audio/video but instead of for saving to file, sending an image to the sender, as well as to another receiving computer. I am guessing this might give you what you want. You do not seem to be naming the tee, and calling it again. gst-launch-0.10 v4l2src ! tee name=tee ! tee. ! queue ! \ video/x-raw-yuv,width=320,height=240 ! ffmpegcolorspace ! ffenc_flv ! \ ffmux_flv ! filesink location=/path/to/file tee. ! queue ! xvimagesink I have not tried the same with the sound, but that should get your video recording and displaying (nota bene: not tested, just an off the top of my head guess) For the sound the use of the tee should do the business. I have not messed with mux(es) yet, so I could have gone wrong there, but the tee stuff should probably resemble the above. You name the tee, and then refer to it later in the commandline by name. Hope this helps, Rohan James Pearson wrote: > So here is my pipeline. The point is to record audio and video and > display it at the same time. > > v4l2src -> tee -> video/x-raw-yuv,width=320,height=240 -> > ffmpegcolorspace -> ffenc_flv -> ffmux_flv -> filesink > -> xvimagesink > > alsasrc -> tee > audioresample -> audio-x-raw-int,rate=44100 - -> > audioconvert -> lame -> ffmux_flv (same mux as above) > -> alsasink > > I have been using this pipeline minus the tee for live playback > (alsasink,xvimagesink) for some time and it is working well. The video > part, even with the live display works ok but when i add in the tee > and alsasink the pipeline gets stuck with nothing going to disk and > the video displayed is stuck. Even just adding the alsasink to the > pipeline without connecting it causes this result. Any suggestions? I > already tried adding queues to almost everything. > > Thanks > > -James- > > > > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From honza at dp.fce.vutbr.cz Tue Sep 29 13:23:16 2009 From: honza at dp.fce.vutbr.cz (Jan Martinek) Date: Tue, 29 Sep 2009 13:23:16 +0200 Subject: [gst-devel] One microphone, recording mono fails Message-ID: <4AC1EE24.4000306@dp.fce.vutbr.cz> Hello, I am unable to record only one channel (mono). I have only one microphone attached to sound card. But if I run this gst-launch gconfaudiosrc ! capsfilter \ caps=audio/x-raw-int,rate=44100,channels=1,depth=16 ! \ wavenc ! filesink location=test.wav then I get this error message: Setting pipeline to PAUSED ... ERROR: Pipeline doesn't want to pause. ERROR: from element /GstPipeline:pipeline0/GstGConfAudioSrc:gconfaudiosrc0/GstBin:bin0/GstAlsaSrc:alsasrc0: Could not negotiate format Additional debug info: gstbasesrc.c(2584): gst_base_src_start (): /GstPipeline:pipeline0/GstGConfAudioSrc:gconfaudiosrc0/GstBin:bin0/GstAlsaSrc:alsasrc0: Check your filtered caps, if any Setting pipeline to NULL ... Freeing pipeline ... It can be solved if I use two channels for recording, so this gst-launch gconfaudiosrc ! capsfilter \ caps=audio/x-raw-int,rate=44100,channels=2,depth=16 ! \ wavenc ! filesink location=test.wav works fine. But I would like to record one channel because further data processing is little easier. Thank you Jan Martinek From virajk at gmail.com Tue Sep 29 13:57:43 2009 From: virajk at gmail.com (Viraj Karandikar) Date: Tue, 29 Sep 2009 17:27:43 +0530 Subject: [gst-devel] One microphone, recording mono fails In-Reply-To: <4AC1EE24.4000306@dp.fce.vutbr.cz> References: <4AC1EE24.4000306@dp.fce.vutbr.cz> Message-ID: Hi, You can capture 2 channels and then use audioconvert to convert it to mono channel. gst-launch gconfaudiosrc ! capsfilter \ caps=audio/x-raw-int,rate=44100,channels=2,depth=16 ! \ audioconvert ! audio/x-raw-int,rate=8000,depth=16,channels=1,width=16,signed=\(boolean\)TRUE,endianness=\(int\)1234 ! wavenc ! filesink location=test.wav regards, Viraj On Tue, Sep 29, 2009 at 4:53 PM, Jan Martinek wrote: > Hello, > > I am unable to record only one channel (mono). I have only one > microphone attached to sound card. But if I run this > > gst-launch gconfaudiosrc ! capsfilter \ > caps=audio/x-raw-int,rate=44100,channels=1,depth=16 ! \ > wavenc ! filesink location=test.wav > > then I get this error message: > > Setting pipeline to PAUSED ... > ERROR: Pipeline doesn't want to pause. > ERROR: from element > > /GstPipeline:pipeline0/GstGConfAudioSrc:gconfaudiosrc0/GstBin:bin0/GstAlsaSrc:alsasrc0: > Could not negotiate format > Additional debug info: > gstbasesrc.c(2584): gst_base_src_start (): > > /GstPipeline:pipeline0/GstGConfAudioSrc:gconfaudiosrc0/GstBin:bin0/GstAlsaSrc:alsasrc0: > Check your filtered caps, if any > Setting pipeline to NULL ... > Freeing pipeline ... > > > It can be solved if I use two channels for recording, so this > > gst-launch gconfaudiosrc ! capsfilter \ > caps=audio/x-raw-int,rate=44100,channels=2,depth=16 ! \ > wavenc ! filesink location=test.wav > > works fine. But I would like to record one channel because further data > processing is little easier. > > Thank you > Jan Martinek > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- - Viraj Reality is merely an illusion, albeit a very persistent one. -------------- next part -------------- An HTML attachment was scrubbed... URL: From honza at dp.fce.vutbr.cz Tue Sep 29 14:25:58 2009 From: honza at dp.fce.vutbr.cz (Jan Martinek) Date: Tue, 29 Sep 2009 14:25:58 +0200 Subject: [gst-devel] One microphone, recording mono fails In-Reply-To: References: <4AC1EE24.4000306@dp.fce.vutbr.cz> Message-ID: <4AC1FCD6.7020800@dp.fce.vutbr.cz> Hi, this is interesting, thank you. But is it a recommended, clean solution or just workaround? I have tried mono recording on a different (older) computer and it works fine there. Relevant line from "lspci" is this: 00:11.5 Multimedia audio controller: VIA Technologies, Inc. VT8233/A/8235/8237 AC97 Audio Controller (rev 50) On the other hand, one channel recording fails on my computer with 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) Does it depend on hardware? Do capabilities differ? I would be surprised, because this seems to me as very basic functionality. Thank you, Jan Martinek On 09/29/2009 01:57 PM, Viraj Karandikar wrote: > Hi, > You can capture 2 channels and then use audioconvert to convert it to > mono channel. > gst-launch gconfaudiosrc ! capsfilter \ > caps=audio/x-raw-int,rate=44100,channels=2,depth=16 ! \ > audioconvert ! > audio/x-raw-int,rate=8000,depth=16,channels=1,width=16,signed=\(boolean\)TRUE,endianness=\(int\)1234 > ! wavenc ! filesink location=test.wav > regards, > Viraj > > On Tue, Sep 29, 2009 at 4:53 PM, Jan Martinek > wrote: > > Hello, > > I am unable to record only one channel (mono). I have only one > microphone attached to sound card. But if I run this > > gst-launch gconfaudiosrc ! capsfilter \ > caps=audio/x-raw-int,rate=44100,channels=1,depth=16 ! \ > wavenc ! filesink location=test.wav > > then I get this error message: > > Setting pipeline to PAUSED ... > ERROR: Pipeline doesn't want to pause. > ERROR: from element > /GstPipeline:pipeline0/GstGConfAudioSrc:gconfaudiosrc0/GstBin:bin0/GstAlsaSrc:alsasrc0: > Could not negotiate format > Additional debug info: > gstbasesrc.c(2584): gst_base_src_start (): > /GstPipeline:pipeline0/GstGConfAudioSrc:gconfaudiosrc0/GstBin:bin0/GstAlsaSrc:alsasrc0: > Check your filtered caps, if any > Setting pipeline to NULL ... > Freeing pipeline ... > > > It can be solved if I use two channels for recording, so this > > gst-launch gconfaudiosrc ! capsfilter \ > caps=audio/x-raw-int,rate=44100,channels=2,depth=16 ! \ > wavenc ! filesink location=test.wav > > works fine. But I would like to record one channel because further data > processing is little easier. > > Thank you > Jan Martinek > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and > stay > ahead of the curve. Join us from November 9-12, 2009. Register > now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > -- > - Viraj > Reality is merely an illusion, albeit a very persistent one. > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > > > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel From virajk at gmail.com Tue Sep 29 15:13:15 2009 From: virajk at gmail.com (Viraj Karandikar) Date: Tue, 29 Sep 2009 18:43:15 +0530 Subject: [gst-devel] One microphone, recording mono fails In-Reply-To: <4AC1FCD6.7020800@dp.fce.vutbr.cz> References: <4AC1EE24.4000306@dp.fce.vutbr.cz> <4AC1FCD6.7020800@dp.fce.vutbr.cz> Message-ID: Hi, I dont see any issue in using audioconvert. Did you see the log by setting debug level to 5? When exactly "Could not negotiate format" error is coming? it might give some clue on whats happening. -Viraj On Tue, Sep 29, 2009 at 5:55 PM, Jan Martinek wrote: > Hi, > > this is interesting, thank you. But is it a recommended, clean solution > or just workaround? > > I have tried mono recording on a different (older) computer and it works > fine there. Relevant line from "lspci" is this: > > 00:11.5 Multimedia audio controller: VIA Technologies, Inc. > VT8233/A/8235/8237 AC97 Audio Controller (rev 50) > > On the other hand, one channel recording fails on my computer with > 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) > > Does it depend on hardware? Do capabilities differ? I would be > surprised, because this seems to me as very basic functionality. > > Thank you, > Jan Martinek > > On 09/29/2009 01:57 PM, Viraj Karandikar wrote: > > Hi, > > You can capture 2 channels and then use audioconvert to convert it to > > mono channel. > > gst-launch gconfaudiosrc ! capsfilter \ > > caps=audio/x-raw-int,rate=44100,channels=2,depth=16 ! \ > > audioconvert ! > > > audio/x-raw-int,rate=8000,depth=16,channels=1,width=16,signed=\(boolean\)TRUE,endianness=\(int\)1234 > > ! wavenc ! filesink location=test.wav > > regards, > > Viraj > > > > On Tue, Sep 29, 2009 at 4:53 PM, Jan Martinek > > wrote: > > > > Hello, > > > > I am unable to record only one channel (mono). I have only one > > microphone attached to sound card. But if I run this > > > > gst-launch gconfaudiosrc ! capsfilter \ > > caps=audio/x-raw-int,rate=44100,channels=1,depth=16 ! \ > > wavenc ! filesink location=test.wav > > > > then I get this error message: > > > > Setting pipeline to PAUSED ... > > ERROR: Pipeline doesn't want to pause. > > ERROR: from element > > > /GstPipeline:pipeline0/GstGConfAudioSrc:gconfaudiosrc0/GstBin:bin0/GstAlsaSrc:alsasrc0: > > Could not negotiate format > > Additional debug info: > > gstbasesrc.c(2584): gst_base_src_start (): > > > /GstPipeline:pipeline0/GstGConfAudioSrc:gconfaudiosrc0/GstBin:bin0/GstAlsaSrc:alsasrc0: > > Check your filtered caps, if any > > Setting pipeline to NULL ... > > Freeing pipeline ... > > > > > > It can be solved if I use two channels for recording, so this > > > > gst-launch gconfaudiosrc ! capsfilter \ > > caps=audio/x-raw-int,rate=44100,channels=2,depth=16 ! \ > > wavenc ! filesink location=test.wav > > > > works fine. But I would like to record one channel because further > data > > processing is little easier. > > > > Thank you > > Jan Martinek > > > > > ------------------------------------------------------------------------------ > > Come build with us! The BlackBerry® Developer Conference in SF, > CA > > is the only developer event you need to attend this year. Jumpstart > your > > developing skills, take BlackBerry mobile applications to market and > > stay > > ahead of the curve. Join us from November 9-12, 2009. Register > > now! > > http://p.sf.net/sfu/devconf > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > -- > > - Viraj > > Reality is merely an illusion, albeit a very persistent one. > > > > > > ------------------------------------------------------------------------ > > > > > ------------------------------------------------------------------------------ > > Come build with us! The BlackBerry® Developer Conference in SF, CA > > is the only developer event you need to attend this year. Jumpstart your > > developing skills, take BlackBerry mobile applications to market and stay > > ahead of the curve. Join us from November 9-12, 2009. Register > now! > > http://p.sf.net/sfu/devconf > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- - Viraj Reality is merely an illusion, albeit a very persistent one. -------------- next part -------------- An HTML attachment was scrubbed... URL: From jam at smru.co.uk Tue Sep 29 15:38:52 2009 From: jam at smru.co.uk (Andy Maginnis) Date: Tue, 29 Sep 2009 14:38:52 +0100 Subject: [gst-devel] One microphone, recording mono fails In-Reply-To: References: <4AC1EE24.4000306@dp.fce.vutbr.cz><4AC1FCD6.7020800@dp.fce.vutbr.cz> Message-ID: I think this is just defaulting to your HW capabilities. Use -v to see whats happening. examples on my machine, gst-launch -v alsasrc device=hw:1 ! audioconvert ! alsasink device=hw:1 gst-launch -v alsasrc device=hw:0 ! audioconvert ! alsasink device=hw:0 produce differing numbers of source channels. I suspect your older computer only has one channel. >LAUNCH 1 /pipeline0/alsasrc0.src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)1 ....... /pipeline0/audioconvert0.src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)2 /pipeline0/audioconvert0.sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)1 /pipeline0/alsasink0.sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)2 >LAUNCH 2 pipeline0/alsasrc0.src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)44100, channels=(int)2 .... /pipeline0/audioconvert0.src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)44100, channels=(int)2 /pipeline0/audioconvert0.sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)44100, channels=(int)2 New clock: GstAudioSrcClock /pipeline0/alsasink0.sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)44100, channels=(int)2/ I can do gst-launch -v alsasrc device=hw:0 ! alsasink device=hw:0 but not gst-launch -v alsasrc device=hw:1 ! alsasink device=hw:1 as I need to modify the number of channels, I need the audioconvert for this one! From: Viraj Karandikar [mailto:virajk at gmail.com] Sent: 29 September 2009 14:13 To: Discussion of the development of GStreamer Subject: Re: [gst-devel] One microphone, recording mono fails Hi, I dont see any issue in using audioconvert. Did you see the log by setting debug level to 5? When exactly "Could not negotiate format" error is coming? it might give some clue on whats happening. -Viraj On Tue, Sep 29, 2009 at 5:55 PM, Jan Martinek wrote: Hi, this is interesting, thank you. But is it a recommended, clean solution or just workaround? I have tried mono recording on a different (older) computer and it works fine there. Relevant line from "lspci" is this: 00:11.5 Multimedia audio controller: VIA Technologies, Inc. VT8233/A/8235/8237 AC97 Audio Controller (rev 50) On the other hand, one channel recording fails on my computer with 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) Does it depend on hardware? Do capabilities differ? I would be surprised, because this seems to me as very basic functionality. Thank you, Jan Martinek On 09/29/2009 01:57 PM, Viraj Karandikar wrote: > Hi, > You can capture 2 channels and then use audioconvert to convert it to > mono channel. > gst-launch gconfaudiosrc ! capsfilter \ > caps=audio/x-raw-int,rate=44100,channels=2,depth=16 ! \ > audioconvert ! > audio/x-raw-int,rate=8000,depth=16,channels=1,width=16,signed=\(boolean\ )TRUE,endianness=\(int\)1234 > ! wavenc ! filesink location=test.wav > regards, > Viraj > > On Tue, Sep 29, 2009 at 4:53 PM, Jan Martinek > wrote: > > Hello, > > I am unable to record only one channel (mono). I have only one > microphone attached to sound card. But if I run this > > gst-launch gconfaudiosrc ! capsfilter \ > caps=audio/x-raw-int,rate=44100,channels=1,depth=16 ! \ > wavenc ! filesink location=test.wav > > then I get this error message: > > Setting pipeline to PAUSED ... > ERROR: Pipeline doesn't want to pause. > ERROR: from element > /GstPipeline:pipeline0/GstGConfAudioSrc:gconfaudiosrc0/GstBin:bin0/GstAl saSrc:alsasrc0: > Could not negotiate format > Additional debug info: > gstbasesrc.c(2584): gst_base_src_start (): > /GstPipeline:pipeline0/GstGConfAudioSrc:gconfaudiosrc0/GstBin:bin0/GstAl saSrc:alsasrc0: > Check your filtered caps, if any > Setting pipeline to NULL ... > Freeing pipeline ... > > > It can be solved if I use two channels for recording, so this > > gst-launch gconfaudiosrc ! capsfilter \ > caps=audio/x-raw-int,rate=44100,channels=2,depth=16 ! \ > wavenc ! filesink location=test.wav > > works fine. But I would like to record one channel because further data > processing is little easier. > > Thank you > Jan Martinek > > ------------------------------------------------------------------------ ------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and > stay > ahead of the curve. Join us from November 9-12, 2009. Register > now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > -- > - Viraj > Reality is merely an illusion, albeit a very persistent one. > > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------ ------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > > > ------------------------------------------------------------------------ > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel ------------------------------------------------------------------------ ------ Come build with us! The BlackBerry® Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9-12, 2009. Register now! http://p.sf.net/sfu/devconf _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -- - Viraj Reality is merely an illusion, albeit a very persistent one. -------------- next part -------------- An HTML attachment was scrubbed... URL: From t.i.m at zen.co.uk Tue Sep 29 15:33:34 2009 From: t.i.m at zen.co.uk (Tim-Philipp =?ISO-8859-1?Q?M=FCller?=) Date: Tue, 29 Sep 2009 14:33:34 +0100 Subject: [gst-devel] One microphone, recording mono fails In-Reply-To: References: <4AC1EE24.4000306@dp.fce.vutbr.cz> <4AC1FCD6.7020800@dp.fce.vutbr.cz> Message-ID: <1254231214.5699.6.camel@zingle> On Tue, 2009-09-29 at 18:43 +0530, Viraj Karandikar wrote: > When exactly "Could not negotiate format" error is coming? it might > give some clue on whats happening. not-negotiated means there was a format incompatibility somewhere. In this case that could be a capsfilter with a format/filter that upstream can't output (e.g. if the source doesn't support recording in mono), or some other element receiving a stream in a format it can't handle. wavenc is particularly picky - you should definitly have an audioconvert in front of wavenc, whatever you do. Something like this should work: gst-launch-0.10 alsasrc ! 'audio/x-raw-int,channels=1;audio/x-raw-int' ! audioconvert ! audio/x-raw-int,channels=1,depth=16 ! wavenc ! filesink location=foo.wav Cheers -Tim From optavas at gmail.com Tue Sep 29 16:41:18 2009 From: optavas at gmail.com (Ottaviano Vasselli) Date: Tue, 29 Sep 2009 16:41:18 +0200 Subject: [gst-devel] unable to load pipeline from XML Message-ID: <6c9581cb0909290741u5c166c3cm8c0b956289f9a4f7@mail.gmail.com> Hi, Stefan Kost wrote: > xml serialisation has some limmitations and is not widley used. In the > above case it might fail to handle the dynamic pads. You could use > G_DEBUG=fatal_warnings gdb --args gst-xmllaunch -vvv saved_pipe.xml > and break on the error. Get the backtrace, figure a fix, report > problem+backtrace+patch to bugzilla. thanks for the addressing... so could you confirm that problem is just loading xml, while gst-launch -o "..." does correctly save? Bye From katcipis at inf.ufsc.br Tue Sep 29 22:01:22 2009 From: katcipis at inf.ufsc.br (Tiago Katcipis) Date: Tue, 29 Sep 2009 17:01:22 -0300 Subject: [gst-devel] SRTP on Gstreamer Message-ID: <60a9403b0909291301x7a14e3a2w3a185a827bdea9ca@mail.gmail.com> Is there any plugin to use SRTP or some ongoing work that i can help? best regards, Katcipis -------------- next part -------------- An HTML attachment was scrubbed... URL: From olivier.crete at collabora.co.uk Tue Sep 29 22:28:34 2009 From: olivier.crete at collabora.co.uk (Olivier =?ISO-8859-1?Q?Cr=EAte?=) Date: Tue, 29 Sep 2009 16:28:34 -0400 Subject: [gst-devel] SRTP on Gstreamer In-Reply-To: <60a9403b0909291301x7a14e3a2w3a185a827bdea9ca@mail.gmail.com> References: <60a9403b0909291301x7a14e3a2w3a185a827bdea9ca@mail.gmail.com> Message-ID: <1254256114.8568.1.camel@TesterTop3.tester.ca> Hi, Our intern Gabriel Millaire is supposed to do it this fall (ie, before the end of the year), but no code has been written yet. On Tue, 2009-09-29 at 17:01 -0300, Tiago Katcipis wrote: > Is there any plugin to use SRTP or some ongoing work that i can help? > > best regards, > Katcipis -- Olivier Cr?te olivier.crete at collabora.co.uk -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: This is a digitally signed message part URL: From katcipis at inf.ufsc.br Tue Sep 29 23:11:12 2009 From: katcipis at inf.ufsc.br (Tiago Katcipis) Date: Tue, 29 Sep 2009 18:11:12 -0300 Subject: [gst-devel] SRTP on Gstreamer In-Reply-To: <1254256114.8568.1.camel@TesterTop3.tester.ca> References: <60a9403b0909291301x7a14e3a2w3a185a827bdea9ca@mail.gmail.com> <1254256114.8568.1.camel@TesterTop3.tester.ca> Message-ID: <60a9403b0909291411u66a733f8ia84e55642573c568@mail.gmail.com> Olivier, I'm working on a solution that will need it, maybe we can work together, or ill start alone, i already written some plugins but they where pretty simple and did very specific things that are completely useless for general purpose, so i didn't even patched it. This srtp plugin will be the first one that can be useful, but i lack in experience, probably the plugin will be a "bad" one :-). I think i may need to start to work on it on the next weeks already. best regards, Katcipis 2009/9/29 Olivier Cr?te > Hi, > > Our intern Gabriel Millaire is supposed to do it this fall (ie, before > the end of the year), but no code has been written yet. > > On Tue, 2009-09-29 at 17:01 -0300, Tiago Katcipis wrote: > > Is there any plugin to use SRTP or some ongoing work that i can help? > > > > best regards, > > Katcipis > > -- > Olivier Cr?te > olivier.crete at collabora.co.uk > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em menos de 5 minutos aparecer? algu?m pra discordar de sua escolha de distribui??o, do particionamento, do gerenciador de janelas, do ambiente de desktop, do editor de textos..." -------------- next part -------------- An HTML attachment was scrubbed... URL: From olivier.crete at collabora.co.uk Tue Sep 29 23:24:47 2009 From: olivier.crete at collabora.co.uk (Olivier =?ISO-8859-1?Q?Cr=EAte?=) Date: Tue, 29 Sep 2009 17:24:47 -0400 Subject: [gst-devel] SRTP on Gstreamer In-Reply-To: <60a9403b0909291411u66a733f8ia84e55642573c568@mail.gmail.com> References: <60a9403b0909291301x7a14e3a2w3a185a827bdea9ca@mail.gmail.com> <1254256114.8568.1.camel@TesterTop3.tester.ca> <60a9403b0909291411u66a733f8ia84e55642573c568@mail.gmail.com> Message-ID: <1254259487.21447.5.camel@TesterTop3.tester.ca> Hi, My plan is to implement SRTP as an independant library and then use is directly from the farsight2 rtp element. Making it a single GStreamer element is not a great idea because it will create a loop with gstrtpbin (I'm also not certain having gstrtpbin as a single element was a great idea at all). So we want to either have multiple gst elements that are inter-connected, or just make it into a non-element library. Maybe do like rtpmanager and have a non-element library that is used by element. Anyway, I'm just thinking aloud here. Olivier On Tue, 2009-09-29 at 18:11 -0300, Tiago Katcipis wrote: > Olivier, > > I'm working on a solution that will need it, maybe we can work > together, or ill start alone, i already written some plugins but they > where pretty simple and did very specific things that are completely > useless for general purpose, so i didn't even patched it. This srtp > plugin will be the first one that can be useful, but i lack in > experience, probably the plugin will be a "bad" one :-). I think i > may need to start to work on it on the next weeks already. > > best regards, > Katcipis > > 2009/9/29 Olivier Cr?te > Hi, > > Our intern Gabriel Millaire is supposed to do it this fall > (ie, before > the end of the year), but no code has been written yet. > > > On Tue, 2009-09-29 at 17:01 -0300, Tiago Katcipis wrote: > > Is there any plugin to use SRTP or some ongoing work that i > can help? > > > > best regards, > > Katcipis > > > -- > Olivier Cr?te > olivier.crete at collabora.co..uk > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference > in SF, CA > is the only developer event you need to attend this year. > Jumpstart your > developing skills, take BlackBerry mobile applications to > market and stay > ahead of the curve. Join us from November 9-12, 2009. > Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > -- > "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta > sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em > menos de 5 minutos aparecer? algu?m pra discordar de sua escolha de > distribui??o, do particionamento, do gerenciador de janelas, do > ambiente de desktop, do editor de textos..." > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -- Olivier Cr?te olivier.crete at collabora.co.uk -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: This is a digitally signed message part URL: From katcipis at inf.ufsc.br Wed Sep 30 01:25:15 2009 From: katcipis at inf.ufsc.br (Tiago Katcipis) Date: Tue, 29 Sep 2009 20:25:15 -0300 Subject: [gst-devel] SRTP on Gstreamer In-Reply-To: <1254259487.21447.5.camel@TesterTop3.tester.ca> References: <60a9403b0909291301x7a14e3a2w3a185a827bdea9ca@mail.gmail.com> <1254256114.8568.1.camel@TesterTop3.tester.ca> <60a9403b0909291411u66a733f8ia84e55642573c568@mail.gmail.com> <1254259487.21447.5.camel@TesterTop3.tester.ca> Message-ID: <60a9403b0909291625n55d5d28bj15791e4c40a2caae@mail.gmail.com> Hi, implement SRTP as an independant library? i was thinking on using libsrtp ( http://srtp.sourceforge.net/srtp.html) to make a new element, i supposed that the independant library part was already done, is there something wrong with this implementation of SRTP? im not very used to gstreamer, if you say that it is not a good idea to make a single gstreamer element, i supose you are right. Im going to study a little more on this :-). best regards, Katcipis 2009/9/29 Olivier Cr?te > Hi, > > My plan is to implement SRTP as an independant library and then use is > directly from the farsight2 rtp element. Making it a single GStreamer > element is not a great idea because it will create a loop with gstrtpbin > (I'm also not certain having gstrtpbin as a single element was a great > idea at all). So we want to either have multiple gst elements that are > inter-connected, or just make it into a non-element library. Maybe do > like rtpmanager and have a non-element library that is used by element. > Anyway, I'm just thinking aloud here. > > Olivier > > On Tue, 2009-09-29 at 18:11 -0300, Tiago Katcipis wrote: > > Olivier, > > > > I'm working on a solution that will need it, maybe we can work > > together, or ill start alone, i already written some plugins but they > > where pretty simple and did very specific things that are completely > > useless for general purpose, so i didn't even patched it. This srtp > > plugin will be the first one that can be useful, but i lack in > > experience, probably the plugin will be a "bad" one :-). I think i > > may need to start to work on it on the next weeks already. > > > > best regards, > > Katcipis > > > > 2009/9/29 Olivier Cr?te > > Hi, > > > > Our intern Gabriel Millaire is supposed to do it this fall > > (ie, before > > the end of the year), but no code has been written yet. > > > > > > On Tue, 2009-09-29 at 17:01 -0300, Tiago Katcipis wrote: > > > Is there any plugin to use SRTP or some ongoing work that i > > can help? > > > > > > best regards, > > > Katcipis > > > > > > -- > > Olivier Cr?te > > olivier.crete at collabora.co..uk > > > > > ------------------------------------------------------------------------------ > > Come build with us! The BlackBerry® Developer Conference > > in SF, CA > > is the only developer event you need to attend this year. > > Jumpstart your > > developing skills, take BlackBerry mobile applications to > > market and stay > > ahead of the curve. Join us from November 9-12, 2009. > > Register now! > > http://p.sf.net/sfu/devconf > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > -- > > "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta > > sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em > > menos de 5 minutos aparecer? algu?m pra discordar de sua escolha de > > distribui??o, do particionamento, do gerenciador de janelas, do > > ambiente de desktop, do editor de textos..." > > > ------------------------------------------------------------------------------ > > Come build with us! The BlackBerry® Developer Conference in SF, CA > > is the only developer event you need to attend this year. Jumpstart your > > developing skills, take BlackBerry mobile applications to market and stay > > ahead of the curve. Join us from November 9-12, 2009. Register > now! > > http://p.sf.net/sfu/devconf > > _______________________________________________ gstreamer-devel mailing > list gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- > Olivier Cr?te > olivier.crete at collabora.co.uk > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em menos de 5 minutos aparecer? algu?m pra discordar de sua escolha de distribui??o, do particionamento, do gerenciador de janelas, do ambiente de desktop, do editor de textos..." -------------- next part -------------- An HTML attachment was scrubbed... URL: From yorgasor at gmail.com Wed Sep 30 01:48:35 2009 From: yorgasor at gmail.com (Ron Yorgason) Date: Tue, 29 Sep 2009 19:48:35 -0400 Subject: [gst-devel] sending raw audio In-Reply-To: <93d1fdd10909240526u413824b0j90c3074cbe81183d@mail.gmail.com> References: <93d1fdd10909231110j181a7bc0w8a2736e8bc6dcdfb@mail.gmail.com> <93d1fdd10909240526u413824b0j90c3074cbe81183d@mail.gmail.com> Message-ID: <93d1fdd10909291648q5e5c0d36y9e7282727e54f9c5@mail.gmail.com> Ok, I'm a little further now. I had an endianness issue that I was able to get around thanks to some other recent discussions on streaming audio in the last couple days. I'm now fighting some serious crackling I get when capturing from the mic and streaming the result across the network. On the capture side, I run this: gst-launch-0.10 -v alsasrc ! audio/x-raw-int,channels=1 ! audioconvert ! audio/x-raw-int,channels=1,depth=16,width=16,rate=24000,endianness=4321 ! rtpL16pay ! udpsink host=192.168.17.81 port=5435 And then the receiver: gst-launch-0.10 -v udpsrc port=5435 caps ="application/x-rtp, media=(string)audio, clock-rate=(int)32000, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, channel-positions=(int)1, payload=(int)96, ssrc=(guint)1168011267, clock-base=(guint)309599748, seqnum-base=(guint)27324, endianness=1234" ! rtpL16depay ! audioconvert ! alsasink sync=false If I capture to a WAV file, and then stream that file across the network, I don't get any crackle. If I do the above capture across the network, I can barely make out the words I'm saying because of the crackle. Now, I'm using a 400Mhz Freescale iMX27 ARM CPU. When I do captures, I get warning messages like these: WARNING: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Can't record audio fast enough Additional debug info: gstbaseaudiosrc.c(805): gst_base_audio_src_create (): /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: dropped 480 samples I think I get more of these messages when I do a straight capture to the network, so it seems likely that the crackling has to do with the dropped samples. The lower the sample rate, the more messages I receive. I can't even capture at 8000 or 16000. I tried to fake my way around this issue by writing to a named pipe as follows: gst-launch-0.10 -v alsasrc ! audio/x-raw-int,channels=1 ! audioconvert ! audio/x-raw-int,channels=1,depth=16,width=16,rate=24000 ! audioconvert ! wavenc ! filesink location=/tmp/whee gst-launch -v --gst-debug=GST_CAT:3 filesrc location=/tmp/whee ! audio/x-raw-int,rate=24000,width=16,depth=16,channels=1,signed=true,endianness=1234 ! audioconvert ! rtpL16pay ! udpsink host=192.168.17.81 port=5435 But this has just as much crackling as sending straight across the network. Does anyone have suggestions on pipeline changes to get rid of the crackling? --Ron On Thu, Sep 24, 2009 at 8:26 AM, Ron Yorgason wrote: > I have a Freescale iMX27. > > --Ron > > On Thu, Sep 24, 2009 at 6:54 AM, Andy Maginnis wrote: >> Ron, >> What ARM are you using? We have a cortex A8, >> inside a OMAP3530 on a Gumstix Overo Water >> a >> >> -----Original Message----- >> From: Ron Yorgason [mailto:yorgasor at gmail.com] >> Sent: 23 September 2009 19:10 >> To: gstreamer-devel at lists.sourceforge.net >> Subject: [gst-devel] sending raw audio >> >> I'm working on an audio/video streaming application on the ARM >> platform. ?We don't seem to have enough CPU horsepower on these boards >> to capture and encode to MP3, so right now we're trying to send raw >> audio. >> >> To capture & stream, I'm using this command: >> >> gst-launch-0.10 -v ?alsasrc ! >> audio/x-raw-int,rate=24000,width=16,depth=16,channels=1,signed=true ! >> audioconvert ! rtpL16pay ?! udpsink host=192.168.17.81 port=5435 >> >> On the receiving side, I have this: >> gst-launch-0.10 -v udpsrc port=5435 caps ="application/x-rtp, >> media=(string)audio, clock-rate=(int)24000, encoding-name=(string)L16, >> encoding-params=(string)1, channels=(int)1, channel-positions=(int)1, >> payload=(int)96, ssrc=(guint)1168011267, clock-base=(guint)309599748, >> seqnum-base=(guint)27324" ?! ? rtpL16depay ! ?audioconvert ! alsasink >> sync=false >> >> When I test this between my desktop & laptop, it works great. ?When I >> go from the ARM board to my desktop, it works ok. ?But when I go from >> the ARM board to another ARM board, or my laptop to the ARM board, I >> hear the speakers turn on, but no sound comes out. ?The output shows >> this: >> >> Setting pipeline to PAUSED ... >> Pipeline is live and does not need PREROLL ... >> Setting pipeline to PLAYING ... >> New clock: GstSystemClock >> /GstPipeline:pipeline0/GstRtpL16Depay:rtpl16depay0.GstPad:src: caps = >> audio/x-raw-int, endianness=(int)4321, signed=(boolean)true, >> width=(int)16, depth=(int)16, rate=(int)24000, channels=(int)1, >> channel-positions=(GstAudioChannelPosition)< >> GST_AUDIO_CHANNEL_POSITION_NONE > >> /GstPipeline:pipeline0/GstRtpL16Depay:rtpl16depay0.GstPad:sink: caps = >> application/x-rtp, media=(string)audio, clock-rate=(int)24000, >> encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, >> channel-positions=(int)1, payload=(int)96, ssrc=(guint)1168011267, >> clock-base=(guint)309599748, seqnum-base=(guint)27324 >> /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps >> = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, >> width=(int)32, depth=(int)32, rate=(int)24000, channels=(int)1, >> channel-positions=(GstAudioChannelPosition)< >> GST_AUDIO_CHANNEL_POSITION_NONE > >> /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps >> = audio/x-raw-int, endianness=(int)4321, signed=(boolean)true, >> width=(int)16, depth=(int)16, rate=(int)24000, channels=(int)1, >> channel-positions=(GstAudioChannelPosition)< >> GST_AUDIO_CHANNEL_POSITION_NONE > >> /GstPipeline:pipeline0/GstAlsaSink:alsasink0.GstPad:sink: caps = >> audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, >> width=(int)32, depth=(int)32, rate=(int)24000, channels=(int)1, >> channel-positions=(GstAudioChannelPosition)< >> GST_AUDIO_CHANNEL_POSITION_NONE > >> >> I couldn't find a good definition of what the >> GST_AUDIO_CHANNEL_POSITION_NONE means, and I wasn't able to set it to >> GST_AUDIO_CHANNEL_POSITION_MONO from the command line (it looks like I >> need to use python or C APIs to do that), but from what I can tell, >> this comes from multichannel support, and I have just specified a >> single audio channel. ?Is this what is preventing me from hearing the >> audio? ?If I capture to a WAV file, and then play it back afterwards, >> it sounds fine. ?So I'm not sure why streaming is failing so badly. >> The playback process also dies with a "Terminated" message within a >> minute or two. >> >> The ARM boards are running gstreamer-0.10.22. ?I see that there's been >> a couple revisions since then, and if I have to upgrade to make it >> work, I will. ?But I'd rather see if there's a way I can make this >> version work. >> >> --Ron >> >> ------------------------------------------------------------------------ >> ------ >> Come build with us! The BlackBerry® Developer Conference in SF, CA >> is the only developer event you need to attend this year. Jumpstart your >> developing skills, take BlackBerry mobile applications to market and >> stay >> ahead of the curve. Join us from November 9-12, 2009. Register >> now! >> http://p.sf.net/sfu/devconf >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> ------------------------------------------------------------------------------ >> Come build with us! The BlackBerry® Developer Conference in SF, CA >> is the only developer event you need to attend this year. Jumpstart your >> developing skills, take BlackBerry mobile applications to market and stay >> ahead of the curve. Join us from November 9-12, 2009. Register now! >> http://p.sf.net/sfu/devconf >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > From olivier.crete at collabora.co.uk Wed Sep 30 01:58:50 2009 From: olivier.crete at collabora.co.uk (Olivier =?ISO-8859-1?Q?Cr=EAte?=) Date: Tue, 29 Sep 2009 19:58:50 -0400 Subject: [gst-devel] SRTP on Gstreamer In-Reply-To: <60a9403b0909291625n55d5d28bj15791e4c40a2caae@mail.gmail.com> References: <60a9403b0909291301x7a14e3a2w3a185a827bdea9ca@mail.gmail.com> <1254256114.8568.1.camel@TesterTop3.tester.ca> <60a9403b0909291411u66a733f8ia84e55642573c568@mail.gmail.com> <1254259487.21447.5.camel@TesterTop3.tester.ca> <60a9403b0909291625n55d5d28bj15791e4c40a2caae@mail.gmail.com> Message-ID: <1254268730.21447.29.camel@TesterTop3.tester.ca> Hi, It looks interesting, although it seems to be no longer developed.. But it may be a good base even if we have to take over developement. I guess we should look into it. Then its just a matter of making the appropriate gst elements. Olivier On Tue, 2009-09-29 at 20:25 -0300, Tiago Katcipis wrote: > Hi, > implement SRTP as an independant library? i was thinking on using > libsrtp (http://srtp.sourceforge.net/srtp.html) to make a new element, > i supposed that the independant library part was already done, is > there something wrong with this implementation of SRTP? > > im not very used to gstreamer, if you say that it is not a good idea > to make a single gstreamer element, i supose you are right. Im going > to study a little more on this :-). > > best regards, > Katcipis > > 2009/9/29 Olivier Cr?te > Hi, > > My plan is to implement SRTP as an independant library and > then use is > directly from the farsight2 rtp element. Making it a single > GStreamer > element is not a great idea because it will create a loop with > gstrtpbin > (I'm also not certain having gstrtpbin as a single element was > a great > idea at all). So we want to either have multiple gst elements > that are > inter-connected, or just make it into a non-element library. > Maybe do > like rtpmanager and have a non-element library that is used by > element. > Anyway, I'm just thinking aloud here. > > Olivier > > > On Tue, 2009-09-29 at 18:11 -0300, Tiago Katcipis wrote: > > Olivier, > > > > I'm working on a solution that will need it, maybe we can > work > > together, or ill start alone, i already written some plugins > but they > > where pretty simple and did very specific things that are > completely > > useless for general purpose, so i didn't even patched it. > This srtp > > plugin will be the first one that can be useful, but i lack > in > > experience, probably the plugin will be a "bad" one :-). I > think i > > may need to start to work on it on the next weeks already. > > > > best regards, > > Katcipis > > > > 2009/9/29 Olivier Cr?te > > Hi, > > > > Our intern Gabriel Millaire is supposed to do it > this fall > > (ie, before > > the end of the year), but no code has been written > yet.. > > > > > > On Tue, 2009-09-29 at 17:01 -0300, Tiago Katcipis > wrote: > > > Is there any plugin to use SRTP or some ongoing > work that i > > can help? > > > > > > best regards, > > > Katcipis > > > > > > -- > > Olivier Cr?te > > olivier.crete at collabora.co..uk > > > > > ------------------------------------------------------------------------------ > > Come build with us! The BlackBerry® Developer > Conference > > in SF, CA > > is the only developer event you need to attend this > year. > > Jumpstart your > > developing skills, take BlackBerry mobile > applications to > > market and stay > > ahead of the curve. Join us from November 9-12, > 2009. > > Register now! > > http://p.sf.net/sfu/devconf > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > -- > > "Se voc? se perder na selva africana, n?o precisa se > desesperar. Basta > > sentar em uma pedra e come?ar a instalar GNU/Linux em seu > laptop. Em > > menos de 5 minutos aparecer? algu?m pra discordar de sua > escolha de > > distribui??o, do particionamento, do gerenciador de janelas, > do > > ambiente de desktop, do editor de textos..." > > > ------------------------------------------------------------------------------ > > Come build with us! The BlackBerry® Developer Conference > in SF, CA > > is the only developer event you need to attend this year. > Jumpstart your > > developing skills, take BlackBerry mobile applications to > market and stay > > ahead of the curve. Join us from November 9-12, 2009. > Register now! > > http://p.sf.net/sfu/devconf > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- > Olivier Cr?te > olivier.crete at collabora.co..uk > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference > in SF, CA > is the only developer event you need to attend this year. > Jumpstart your > developing skills, take BlackBerry mobile applications to > market and stay > ahead of the curve. Join us from November 9-12, 2009. > Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > -- > "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta > sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em > menos de 5 minutos aparecer? algu?m pra discordar de sua escolha de > distribui??o, do particionamento, do gerenciador de janelas, do > ambiente de desktop, do editor de textos..." > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -- Olivier Cr?te olivier.crete at collabora.co.uk -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: This is a digitally signed message part URL: From katcipis at inf.ufsc.br Wed Sep 30 03:40:57 2009 From: katcipis at inf.ufsc.br (Tiago Katcipis) Date: Tue, 29 Sep 2009 22:40:57 -0300 Subject: [gst-devel] SRTP on Gstreamer In-Reply-To: <1254268730.21447.29.camel@TesterTop3.tester.ca> References: <60a9403b0909291301x7a14e3a2w3a185a827bdea9ca@mail.gmail.com> <1254256114.8568.1.camel@TesterTop3.tester.ca> <60a9403b0909291411u66a733f8ia84e55642573c568@mail.gmail.com> <1254259487.21447.5.camel@TesterTop3.tester.ca> <60a9403b0909291625n55d5d28bj15791e4c40a2caae@mail.gmail.com> <1254268730.21447.29.camel@TesterTop3.tester.ca> Message-ID: <60a9403b0909291840j4126f204hda25462362066e14@mail.gmail.com> I think the same, even taking over may be a good idea. The lib seens to work well and i was able to cross-compile it with mingw, so even making a gstelement to work on windows may not be a problem (it seens to have suport for power pc too). But i have no idea of how is the source code, if it is well designed..etc. But i have heard that it is pretty easy to use. best regards, Katcipis 2009/9/29 Olivier Cr?te > Hi, > > It looks interesting, although it seems to be no longer developed.. But > it may be a good base even if we have to take over developement. I guess > we should look into it. Then its just a matter of making the appropriate > gst elements. > > Olivier > > On Tue, 2009-09-29 at 20:25 -0300, Tiago Katcipis wrote: > > Hi, > > implement SRTP as an independant library? i was thinking on using > > libsrtp (http://srtp.sourceforge.net/srtp.html) to make a new element, > > i supposed that the independant library part was already done, is > > there something wrong with this implementation of SRTP? > > > > im not very used to gstreamer, if you say that it is not a good idea > > to make a single gstreamer element, i supose you are right. Im going > > to study a little more on this :-). > > > > best regards, > > Katcipis > > > > 2009/9/29 Olivier Cr?te > > Hi, > > > > My plan is to implement SRTP as an independant library and > > then use is > > directly from the farsight2 rtp element. Making it a single > > GStreamer > > element is not a great idea because it will create a loop with > > gstrtpbin > > (I'm also not certain having gstrtpbin as a single element was > > a great > > idea at all). So we want to either have multiple gst elements > > that are > > inter-connected, or just make it into a non-element library. > > Maybe do > > like rtpmanager and have a non-element library that is used by > > element. > > Anyway, I'm just thinking aloud here. > > > > Olivier > > > > > > On Tue, 2009-09-29 at 18:11 -0300, Tiago Katcipis wrote: > > > Olivier, > > > > > > I'm working on a solution that will need it, maybe we can > > work > > > together, or ill start alone, i already written some plugins > > but they > > > where pretty simple and did very specific things that are > > completely > > > useless for general purpose, so i didn't even patched it. > > This srtp > > > plugin will be the first one that can be useful, but i lack > > in > > > experience, probably the plugin will be a "bad" one :-). I > > think i > > > may need to start to work on it on the next weeks already. > > > > > > best regards, > > > Katcipis > > > > > > 2009/9/29 Olivier Cr?te > > > Hi, > > > > > > Our intern Gabriel Millaire is supposed to do it > > this fall > > > (ie, before > > > the end of the year), but no code has been written > > yet.. > > > > > > > > > On Tue, 2009-09-29 at 17:01 -0300, Tiago Katcipis > > wrote: > > > > Is there any plugin to use SRTP or some ongoing > > work that i > > > can help? > > > > > > > > best regards, > > > > Katcipis > > > > > > > > > -- > > > Olivier Cr?te > > > olivier.crete at collabora.co..uk > > > > > > > > > ------------------------------------------------------------------------------ > > > Come build with us! The BlackBerry® Developer > > Conference > > > in SF, CA > > > is the only developer event you need to attend this > > year. > > > Jumpstart your > > > developing skills, take BlackBerry mobile > > applications to > > > market and stay > > > ahead of the curve. Join us from November 9-12, > > 2009. > > > Register now! > > > http://p.sf.net/sfu/devconf > > > _______________________________________________ > > > gstreamer-devel mailing list > > > gstreamer-devel at lists.sourceforge.net > > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > > > > > > -- > > > "Se voc? se perder na selva africana, n?o precisa se > > desesperar. Basta > > > sentar em uma pedra e come?ar a instalar GNU/Linux em seu > > laptop. Em > > > menos de 5 minutos aparecer? algu?m pra discordar de sua > > escolha de > > > distribui??o, do particionamento, do gerenciador de janelas, > > do > > > ambiente de desktop, do editor de textos..." > > > > > > ------------------------------------------------------------------------------ > > > Come build with us! The BlackBerry® Developer Conference > > in SF, CA > > > is the only developer event you need to attend this year. > > Jumpstart your > > > developing skills, take BlackBerry mobile applications to > > market and stay > > > ahead of the curve. Join us from November 9-12, 2009. > > Register now! > > > http://p.sf.net/sfu/devconf > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- > > Olivier Cr?te > > olivier.crete at collabora.co..uk > > > > > > > ------------------------------------------------------------------------------ > > Come build with us! The BlackBerry® Developer Conference > > in SF, CA > > is the only developer event you need to attend this year. > > Jumpstart your > > developing skills, take BlackBerry mobile applications to > > market and stay > > ahead of the curve. Join us from November 9-12, 2009. > > Register now! > > http://p.sf.net/sfu/devconf > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > -- > > "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta > > sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em > > menos de 5 minutos aparecer? algu?m pra discordar de sua escolha de > > distribui??o, do particionamento, do gerenciador de janelas, do > > ambiente de desktop, do editor de textos..." > > > ------------------------------------------------------------------------------ > > Come build with us! The BlackBerry® Developer Conference in SF, CA > > is the only developer event you need to attend this year. Jumpstart your > > developing skills, take BlackBerry mobile applications to market and stay > > ahead of the curve. Join us from November 9-12, 2009. Register > now! > > http://p.sf.net/sfu/devconf > > _______________________________________________ gstreamer-devel mailing > list gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- > Olivier Cr?te > olivier.crete at collabora.co.uk > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em menos de 5 minutos aparecer? algu?m pra discordar de sua escolha de distribui??o, do particionamento, do gerenciador de janelas, do ambiente de desktop, do editor de textos..." -------------- next part -------------- An HTML attachment was scrubbed... URL: From dxssx.dxssx at gmail.com Wed Sep 30 06:18:47 2009 From: dxssx.dxssx at gmail.com (dxssx) Date: Wed, 30 Sep 2009 12:18:47 +0800 Subject: [gst-devel] qtdemux bug? In-Reply-To: <1254212488.5699.1.camel@zingle> References: <1254212488.5699.1.camel@zingle> Message-ID: thanks 2009/9/29 Tim-Philipp M?ller > On Tue, 2009-09-29 at 11:36 +0800, dxssx wrote: > > > Pipeline "filesrc location=xxx.mp4 ! qtdemux name=demux > > demux.audio_00 ! queue ! filesink location=a.aac demux.video_00 ! > > queue ! filesink location=a.h264" works well, but "filesrc > > location=xxx.mp4 ! qtdemux name=demux demux.audio_00 ! filesink > > location=a.aac demux.video_00 ! filesink location=a.h264" does not > > work. > > > > > > It seems to be a dead lock in basesink which is waiting for a play > > event. > > > > > > Does it necessary to add a queue in pipeline when demuxing? > > Yes, you need to add queues, otherwise the pipeline can't preroll > because the first sink to get data will block and control won't be given > back to qtdemux to push data to the other sink (which will then wait for > data forever and never preroll). > > Cheers > -Tim > > > > > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -------------- next part -------------- An HTML attachment was scrubbed... URL: From havard.graff at tandberg.com Wed Sep 30 06:48:43 2009 From: havard.graff at tandberg.com (=?iso-8859-1?Q?H=E5vard_Graff?=) Date: Wed, 30 Sep 2009 06:48:43 +0200 Subject: [gst-devel] SRTP on Gstreamer In-Reply-To: <60a9403b0909291840j4126f204hda25462362066e14@mail.gmail.com> References: <60a9403b0909291301x7a14e3a2w3a185a827bdea9ca@mail.gmail.com><1254256114.8568.1.camel@TesterTop3.tester.ca><60a9403b0909291411u66a733f8ia84e55642573c568@mail.gmail.com><1254259487.21447.5.camel@TesterTop3.tester.ca><60a9403b0909291625n55d5d28bj15791e4c40a2caae@mail.gmail.com><1254268730.21447.29.camel@TesterTop3.tester.ca> <60a9403b0909291840j4126f204hda25462362066e14@mail.gmail.com> Message-ID: <9F6ACAE02B6DD040A1E259977622CFDB0663DFA2@oslexcp1.eu.tandberg.int> Hi, We have SRTP in our application, and we made a SRTP encoder and decoder element. Works like a charm. You need some events / messages / errors to tell the application about stuff like invalid keys and replay-attacks, and to be able to set the key on the elements (properties), but that is trivial. Regards, H?vard ________________________________ From: Tiago Katcipis [mailto:katcipis at inf.ufsc.br] Sent: 30. september 2009 03:41 To: Discussion of the development of GStreamer Cc: gabriel.millaire at collabora.co.uk Subject: Re: [gst-devel] SRTP on Gstreamer I think the same, even taking over may be a good idea. The lib seens to work well and i was able to cross-compile it with mingw, so even making a gstelement to work on windows may not be a problem (it seens to have suport for power pc too). But i have no idea of how is the source code, if it is well designed..etc. But i have heard that it is pretty easy to use. best regards, Katcipis 2009/9/29 Olivier Cr?te Hi, It looks interesting, although it seems to be no longer developed.. But it may be a good base even if we have to take over developement. I guess we should look into it. Then its just a matter of making the appropriate gst elements. Olivier On Tue, 2009-09-29 at 20:25 -0300, Tiago Katcipis wrote: > Hi, > implement SRTP as an independant library? i was thinking on using > libsrtp (http://srtp.sourceforge.net/srtp.html) to make a new element, > i supposed that the independant library part was already done, is > there something wrong with this implementation of SRTP? > > im not very used to gstreamer, if you say that it is not a good idea > to make a single gstreamer element, i supose you are right. Im going > to study a little more on this :-). > > best regards, > Katcipis > > 2009/9/29 Olivier Cr?te > Hi, > > My plan is to implement SRTP as an independant library and > then use is > directly from the farsight2 rtp element. Making it a single > GStreamer > element is not a great idea because it will create a loop with > gstrtpbin > (I'm also not certain having gstrtpbin as a single element was > a great > idea at all). So we want to either have multiple gst elements > that are > inter-connected, or just make it into a non-element library. > Maybe do > like rtpmanager and have a non-element library that is used by > element. > Anyway, I'm just thinking aloud here. > > Olivier > > > On Tue, 2009-09-29 at 18:11 -0300, Tiago Katcipis wrote: > > Olivier, > > > > I'm working on a solution that will need it, maybe we can > work > > together, or ill start alone, i already written some plugins > but they > > where pretty simple and did very specific things that are > completely > > useless for general purpose, so i didn't even patched it. > This srtp > > plugin will be the first one that can be useful, but i lack > in > > experience, probably the plugin will be a "bad" one :-). I > think i > > may need to start to work on it on the next weeks already. > > > > best regards, > > Katcipis > > > > 2009/9/29 Olivier Cr?te > > Hi, > > > > Our intern Gabriel Millaire is supposed to do it > this fall > > (ie, before > > the end of the year), but no code has been written > yet.. > > > > > > On Tue, 2009-09-29 at 17:01 -0300, Tiago Katcipis > wrote: > > > Is there any plugin to use SRTP or some ongoing > work that i > > can help? > > > > > > best regards, > > > Katcipis > > > > > > -- > > Olivier Cr?te > > olivier.crete at collabora.co..uk > > > > > ------------------------------------------------------------------------------ > > Come build with us! The BlackBerry® Developer > Conference > > in SF, CA > > is the only developer event you need to attend this > year. > > Jumpstart your > > developing skills, take BlackBerry mobile > applications to > > market and stay > > ahead of the curve. Join us from November 9-12, > 2009. > > Register now! > > http://p.sf.net/sfu/devconf > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > -- > > "Se voc? se perder na selva africana, n?o precisa se > desesperar. Basta > > sentar em uma pedra e come?ar a instalar GNU/Linux em seu > laptop. Em > > menos de 5 minutos aparecer? algu?m pra discordar de sua > escolha de > > distribui??o, do particionamento, do gerenciador de janelas, > do > > ambiente de desktop, do editor de textos..." > > > ------------------------------------------------------------------------------ > > Come build with us! The BlackBerry® Developer Conference > in SF, CA > > is the only developer event you need to attend this year. > Jumpstart your > > developing skills, take BlackBerry mobile applications to > market and stay > > ahead of the curve. Join us from November 9-12, 2009. > Register now! > > http://p.sf.net/sfu/devconf > > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- > Olivier Cr?te > olivier.crete at collabora.co..uk > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference > in SF, CA > is the only developer event you need to attend this year. > Jumpstart your > developing skills, take BlackBerry mobile applications to > market and stay > ahead of the curve. Join us from November 9-12, 2009. > Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > -- > "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta > sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em > menos de 5 minutos aparecer? algu?m pra discordar de sua escolha de > distribui??o, do particionamento, do gerenciador de janelas, do > ambiente de desktop, do editor de textos..." > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -- Olivier Cr?te olivier.crete at collabora.co.uk ------------------------------------------------------------------------------ Come build with us! The BlackBerry® Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9-12, 2009. Register now! http://p.sf.net/sfu/devconf _______________________________________________ gstreamer-devel mailing list gstreamer-devel at lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/gstreamer-devel -- "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em menos de 5 minutos aparecer? algu?m pra discordar de sua escolha de distribui??o, do particionamento, do gerenciador de janelas, do ambiente de desktop, do editor de textos..." -------------- next part -------------- An HTML attachment was scrubbed... URL: From pratheesh at ti.com Wed Sep 30 09:11:30 2009 From: pratheesh at ti.com (TK, Pratheesh Gangadhar) Date: Wed, 30 Sep 2009 12:41:30 +0530 Subject: [gst-devel] Patch for running gst-plugins-gl in beagleboard Message-ID: Hi, I am using gst-plugins-gl (commit id: 18f5c4875006606b28aa9aa366abbc5dd1e16b60) in beagleboard (OMAP3). While running this I saw this error GL_FRAMEBUFFER_INCOMPLETE_DIMENSIONS and abort during init. After some digging I think the failure is due to this As per http://www.khronos.org/opengles/sdk/docs/man/glRenderbufferStorage.xml glRenderbufferStorage takes only one of these as internalformat (GL_RGBA4,GL_RGB565,GL_RGB5_A1,GL_DEPTH_COMPONENT16, GL_STENCIL_INDEX8) so IMG driver implementation for SGX530 is complying to this whereas gst-plugins-gl uses GL_DEPTH_COMPONENT for GLES now. With below patch I can run filtercube plugin in beagleboard. --- /tmp/gstgldisplay.c 2009-09-28 00:49:48.000000000 +0530 +++ git/gst-libs/gst/gl/gstgldisplay.c 2009-09-30 12:32:53.000000000 +0530 @@ -1172,9 +1172,13 @@ gst_gl_display_thread_init_download (Gst glGenRenderbuffersEXT (1, &display->download_depth_buffer); glBindRenderbufferEXT (GL_RENDERBUFFER_EXT, display->download_depth_buffer); +#ifndef OPENGL_ES2 glRenderbufferStorageEXT (GL_RENDERBUFFER_EXT, GL_DEPTH_COMPONENT, display->download_width, display->download_height); - +#else + glRenderbufferStorageEXT (GL_RENDERBUFFER_EXT, GL_DEPTH_COMPONENT16, + display->download_width, display->download_height); +#endif //setup a first texture to render to glGenTextures (1, &display->download_texture); glBindTexture (GL_TEXTURE_RECTANGLE_ARB, display->download_texture); @@ -1536,8 +1540,13 @@ gst_gl_display_thread_gen_fbo (GstGLDisp //setup the render buffer for depth glGenRenderbuffersEXT (1, &display->generated_depth_buffer); glBindRenderbufferEXT (GL_RENDERBUFFER_EXT, display->generated_depth_buffer); +#ifndef OPENGL_ES2 glRenderbufferStorageEXT (GL_RENDERBUFFER_EXT, GL_DEPTH_COMPONENT, - display->gen_fbo_width, display->gen_fbo_height); + display->download_width, display->download_height); +#else + glRenderbufferStorageEXT (GL_RENDERBUFFER_EXT, GL_DEPTH_COMPONENT16, + display->download_width, display->download_height); +#endif //setup a texture to render to glGenTextures (1, &fake_texture); @@ -2422,8 +2431,13 @@ gst_gl_display_thread_init_upload_fbo (G //setup the render buffer for depth glGenRenderbuffersEXT (1, &display->upload_depth_buffer); glBindRenderbufferEXT (GL_RENDERBUFFER_EXT, display->upload_depth_buffer); +#ifndef OPENGL_ES2 glRenderbufferStorageEXT (GL_RENDERBUFFER_EXT, GL_DEPTH_COMPONENT, - display->upload_width, display->upload_height); + display->download_width, display->download_height); +#else + glRenderbufferStorageEXT (GL_RENDERBUFFER_EXT, GL_DEPTH_COMPONENT16, + display->download_width, display->download_height); +#endif //a fake texture is attached to the upload FBO (cannot init without it) glGenTextures (1, &fake_texture); Regards, Pratheesh From gdevel at clixxun.com Wed Sep 30 09:22:27 2009 From: gdevel at clixxun.com (Roland Peffer) Date: Wed, 30 Sep 2009 09:22:27 +0200 Subject: [gst-devel] gstreamer.freedesktop.org down? In-Reply-To: References: Message-ID: <8DF5348B-1621-45EC-BD23-9292CA3A46F0@clixxun.com> Good Morning, I can not reach gstreamer.freedesktop.org site. Is it down? Regards, Roland From smcnam at gmail.com Wed Sep 30 09:56:10 2009 From: smcnam at gmail.com (Sean McNamara) Date: Wed, 30 Sep 2009 03:56:10 -0400 Subject: [gst-devel] gstreamer.freedesktop.org down? In-Reply-To: <8DF5348B-1621-45EC-BD23-9292CA3A46F0@clixxun.com> References: <8DF5348B-1621-45EC-BD23-9292CA3A46F0@clixxun.com> Message-ID: <74eb1fe20909300056r618b0799kf2f150b96ed1d8ba@mail.gmail.com> Looks to be that *.freedesktop.org is down, even cgit and www :( Confirmed locally, and these guys seem to agree: http://downforeveryoneorjustme.com/ On Wed, Sep 30, 2009 at 3:22 AM, Roland Peffer wrote: > Good Morning, > > I can not reach gstreamer.freedesktop.org site. Is it down? > > Regards, > > Roland > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > From trungnthut at gmail.com Wed Sep 30 11:24:39 2009 From: trungnthut at gmail.com (=?UTF-8?B?VGjDoG5oIFRydW5nIE5ndXnhu4Vu?=) Date: Wed, 30 Sep 2009 16:24:39 +0700 Subject: [gst-devel] runtime parameter adjusting Message-ID: <5b9896820909300224k599ce8f8pf15447594e7323c1@mail.gmail.com> Hi all, Is there any one know if there's existing elements that can adjust parameters at runtime. For examples: adjusting video framerate, bitrate or something like that. Also, if you know or got any ideas about that, please tell me. Thanks -- Cheers ! trungnt -------------- next part -------------- An HTML attachment was scrubbed... URL: From julien.isorce at gmail.com Wed Sep 30 11:35:06 2009 From: julien.isorce at gmail.com (Julien Isorce) Date: Wed, 30 Sep 2009 11:35:06 +0200 Subject: [gst-devel] Patch for running gst-plugins-gl in beagleboard In-Reply-To: References: Message-ID: <180a127d0909300235r51eba0deyb52761a6ddd5b05d@mail.gmail.com> Hi, Nice ! I think you also resolved this: https://bugzilla.gnome.org/show_bug.cgi?id=593786 Could you attach your patch to this bug ? Sincerely Julien 2009/9/30 TK, Pratheesh Gangadhar > Hi, > > I am using gst-plugins-gl (commit id: > 18f5c4875006606b28aa9aa366abbc5dd1e16b60) in beagleboard (OMAP3). While > running this I saw this error GL_FRAMEBUFFER_INCOMPLETE_DIMENSIONS and abort > during init. After some digging I think the failure is due to this > > As per > http://www.khronos.org/opengles/sdk/docs/man/glRenderbufferStorage.xml > > glRenderbufferStorage takes only one of these as internalformat > (GL_RGBA4,GL_RGB565,GL_RGB5_A1,GL_DEPTH_COMPONENT16, GL_STENCIL_INDEX8) so > IMG driver implementation for SGX530 is complying to this whereas > gst-plugins-gl uses GL_DEPTH_COMPONENT for GLES now. > > > With below patch I can run filtercube plugin in beagleboard. > --- /tmp/gstgldisplay.c 2009-09-28 00:49:48.000000000 +0530 > +++ git/gst-libs/gst/gl/gstgldisplay.c 2009-09-30 12:32:53.000000000 +0530 > @@ -1172,9 +1172,13 @@ gst_gl_display_thread_init_download (Gst > glGenRenderbuffersEXT (1, &display->download_depth_buffer); > glBindRenderbufferEXT (GL_RENDERBUFFER_EXT, > display->download_depth_buffer); > +#ifndef OPENGL_ES2 > glRenderbufferStorageEXT (GL_RENDERBUFFER_EXT, GL_DEPTH_COMPONENT, > display->download_width, display->download_height); > - > +#else > + glRenderbufferStorageEXT (GL_RENDERBUFFER_EXT, > GL_DEPTH_COMPONENT16, > + display->download_width, display->download_height); > +#endif > //setup a first texture to render to > glGenTextures (1, &display->download_texture); > glBindTexture (GL_TEXTURE_RECTANGLE_ARB, > display->download_texture); > @@ -1536,8 +1540,13 @@ gst_gl_display_thread_gen_fbo (GstGLDisp > //setup the render buffer for depth > glGenRenderbuffersEXT (1, &display->generated_depth_buffer); > glBindRenderbufferEXT (GL_RENDERBUFFER_EXT, > display->generated_depth_buffer); > +#ifndef OPENGL_ES2 > glRenderbufferStorageEXT (GL_RENDERBUFFER_EXT, GL_DEPTH_COMPONENT, > - display->gen_fbo_width, display->gen_fbo_height); > + display->download_width, display->download_height); > +#else > + glRenderbufferStorageEXT (GL_RENDERBUFFER_EXT, GL_DEPTH_COMPONENT16, > + display->download_width, display->download_height); > +#endif > > //setup a texture to render to > glGenTextures (1, &fake_texture); > @@ -2422,8 +2431,13 @@ gst_gl_display_thread_init_upload_fbo (G > //setup the render buffer for depth > glGenRenderbuffersEXT (1, &display->upload_depth_buffer); > glBindRenderbufferEXT (GL_RENDERBUFFER_EXT, > display->upload_depth_buffer); > +#ifndef OPENGL_ES2 > glRenderbufferStorageEXT (GL_RENDERBUFFER_EXT, GL_DEPTH_COMPONENT, > - display->upload_width, display->upload_height); > + display->download_width, display->download_height); > +#else > + glRenderbufferStorageEXT (GL_RENDERBUFFER_EXT, GL_DEPTH_COMPONENT16, > + display->download_width, display->download_height); > +#endif > > //a fake texture is attached to the upload FBO (cannot init without it) > glGenTextures (1, &fake_texture); > > Regards, > Pratheesh > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From 4ernov at gmail.com Wed Sep 30 12:04:45 2009 From: 4ernov at gmail.com (Alexey Chernov) Date: Wed, 30 Sep 2009 14:04:45 +0400 Subject: [gst-devel] Appsink output with audio/x-raw-float caps Message-ID: <200909301404.45259.4ernov@gmail.com> Hello, I've got two more questions concerning appsink during futher development of my sound editor project: 1. Is it possible that appsink produces NaNs in its output with "audio/x-raw- float,width=32,endianness=1234" caps? Or maybe I illegally convert to float from the output buffer of bytes (I just read the stored buffer as vector of floats with length divided by 4).. There're so many NaNs in the output flow after that and the question is whether it is normal for appsink output or it's my wrong convertion. 2. Could you please help me with any links to audio/x-raw-float format reference? Especially the maths of storing sample in floating point value and the way these values are normalized (I need to visualize the wave so it's important for me to know). I did my best to find it in the web but almost no results.. Thank you very much in advance. From katcipis at inf.ufsc.br Wed Sep 30 12:55:46 2009 From: katcipis at inf.ufsc.br (Tiago Katcipis) Date: Wed, 30 Sep 2009 07:55:46 -0300 Subject: [gst-devel] SRTP on Gstreamer In-Reply-To: <9F6ACAE02B6DD040A1E259977622CFDB0663DFA2@oslexcp1.eu.tandberg.int> References: <60a9403b0909291301x7a14e3a2w3a185a827bdea9ca@mail.gmail.com> <1254256114.8568.1.camel@TesterTop3.tester.ca> <60a9403b0909291411u66a733f8ia84e55642573c568@mail.gmail.com> <1254259487.21447.5.camel@TesterTop3.tester.ca> <60a9403b0909291625n55d5d28bj15791e4c40a2caae@mail.gmail.com> <1254268730.21447.29.camel@TesterTop3.tester.ca> <60a9403b0909291840j4126f204hda25462362066e14@mail.gmail.com> <9F6ACAE02B6DD040A1E259977622CFDB0663DFA2@oslexcp1.eu.tandberg.int> Message-ID: <60a9403b0909300355q3bb13ae1y8fbf46d95b13dc65@mail.gmail.com> Hi, Since it works fine cant you patch it up and propose to add on gstreamer? Then we all can use it and make it work even better. You used libsrtp to implement this encoder / decoder ? best regards, Katcipis On Wed, Sep 30, 2009 at 1:48 AM, H?vard Graff wrote: > Hi, > > > > We have SRTP in our application, and we made a SRTP encoder and decoder > element. Works like a charm. You need some events / messages / errors to > tell the application about stuff like invalid keys and replay-attacks, and > to be able to set the key on the elements (properties), but that is trivial. > > > > Regards, > > H?vard > > > > > ------------------------------ > > *From:* Tiago Katcipis [mailto:katcipis at inf.ufsc.br] > *Sent:* 30. september 2009 03:41 > *To:* Discussion of the development of GStreamer > *Cc:* gabriel.millaire at collabora.co.uk > *Subject:* Re: [gst-devel] SRTP on Gstreamer > > > > I think the same, even taking over may be a good idea. The lib seens to > work well and i was able to cross-compile it with mingw, so even making a > gstelement to work on windows may not be a problem (it seens to have suport > for power pc too). But i have no idea of how is the source code, if it is > well designed..etc. But i have heard that it is pretty easy to use. > > best regards, > Katcipis > > 2009/9/29 Olivier Cr?te > > Hi, > > It looks interesting, although it seems to be no longer developed.. But > it may be a good base even if we have to take over developement. I guess > we should look into it. Then its just a matter of making the appropriate > gst elements. > > Olivier > > > On Tue, 2009-09-29 at 20:25 -0300, Tiago Katcipis wrote: > > Hi, > > implement SRTP as an independant library? i was thinking on using > > libsrtp (http://srtp.sourceforge.net/srtp.html) to make a new element, > > i supposed that the independant library part was already done, is > > there something wrong with this implementation of SRTP? > > > > im not very used to gstreamer, if you say that it is not a good idea > > to make a single gstreamer element, i supose you are right. Im going > > to study a little more on this :-). > > > > best regards, > > Katcipis > > > > 2009/9/29 Olivier Cr?te > > Hi, > > > > My plan is to implement SRTP as an independant library and > > then use is > > directly from the farsight2 rtp element. Making it a single > > GStreamer > > element is not a great idea because it will create a loop with > > gstrtpbin > > (I'm also not certain having gstrtpbin as a single element was > > a great > > idea at all). So we want to either have multiple gst elements > > that are > > inter-connected, or just make it into a non-element library. > > Maybe do > > like rtpmanager and have a non-element library that is used by > > element. > > Anyway, I'm just thinking aloud here. > > > > Olivier > > > > > > On Tue, 2009-09-29 at 18:11 -0300, Tiago Katcipis wrote: > > > Olivier, > > > > > > I'm working on a solution that will need it, maybe we can > > work > > > together, or ill start alone, i already written some plugins > > but they > > > where pretty simple and did very specific things that are > > completely > > > useless for general purpose, so i didn't even patched it. > > This srtp > > > plugin will be the first one that can be useful, but i lack > > in > > > experience, probably the plugin will be a "bad" one :-). I > > think i > > > may need to start to work on it on the next weeks already. > > > > > > best regards, > > > Katcipis > > > > > > 2009/9/29 Olivier Cr?te > > > Hi, > > > > > > Our intern Gabriel Millaire is supposed to do it > > this fall > > > (ie, before > > > the end of the year), but no code has been written > > yet.. > > > > > > > > > On Tue, 2009-09-29 at 17:01 -0300, Tiago Katcipis > > wrote: > > > > Is there any plugin to use SRTP or some ongoing > > work that i > > > can help? > > > > > > > > best regards, > > > > Katcipis > > > > > > > > > -- > > > Olivier Cr?te > > > olivier.crete at collabora.co..uk > > > > > > > > > ------------------------------------------------------------------------------ > > > Come build with us! The BlackBerry® Developer > > Conference > > > in SF, CA > > > is the only developer event you need to attend this > > year. > > > Jumpstart your > > > developing skills, take BlackBerry mobile > > applications to > > > market and stay > > > ahead of the curve. Join us from November 9-12, > > 2009. > > > Register now! > > > http://p.sf.net/sfu/devconf > > > _______________________________________________ > > > gstreamer-devel mailing list > > > gstreamer-devel at lists.sourceforge.net > > > > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > > > > > > -- > > > "Se voc? se perder na selva africana, n?o precisa se > > desesperar. Basta > > > sentar em uma pedra e come?ar a instalar GNU/Linux em seu > > laptop. Em > > > menos de 5 minutos aparecer? algu?m pra discordar de sua > > escolha de > > > distribui??o, do particionamento, do gerenciador de janelas, > > do > > > ambiente de desktop, do editor de textos..." > > > > > > ------------------------------------------------------------------------------ > > > Come build with us! The BlackBerry® Developer Conference > > in SF, CA > > > is the only developer event you need to attend this year. > > Jumpstart your > > > developing skills, take BlackBerry mobile applications to > > market and stay > > > ahead of the curve. Join us from November 9-12, 2009. > > Register now! > > > http://p.sf.net/sfu/devconf > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- > > Olivier Cr?te > > olivier.crete at collabora.co..uk > > > > > > > ------------------------------------------------------------------------------ > > Come build with us! The BlackBerry® Developer Conference > > in SF, CA > > is the only developer event you need to attend this year. > > Jumpstart your > > developing skills, take BlackBerry mobile applications to > > market and stay > > ahead of the curve. Join us from November 9-12, 2009. > > Register now! > > http://p.sf.net/sfu/devconf > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > > > > > > -- > > "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta > > sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em > > menos de 5 minutos aparecer? algu?m pra discordar de sua escolha de > > distribui??o, do particionamento, do gerenciador de janelas, do > > ambiente de desktop, do editor de textos..." > > > ------------------------------------------------------------------------------ > > Come build with us! The BlackBerry® Developer Conference in SF, CA > > is the only developer event you need to attend this year. Jumpstart your > > developing skills, take BlackBerry mobile applications to market and stay > > ahead of the curve. Join us from November 9-12, 2009. Register > now! > > http://p.sf.net/sfu/devconf > > _______________________________________________ gstreamer-devel mailing > list gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- > Olivier Cr?te > olivier.crete at collabora.co.uk > > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > > -- > "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta > sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em menos > de 5 minutos aparecer? algu?m pra discordar de sua escolha de distribui??o, > do particionamento, do gerenciador de janelas, do ambiente de desktop, do > editor de textos..." > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- "Se voc? se perder na selva africana, n?o precisa se desesperar. Basta sentar em uma pedra e come?ar a instalar GNU/Linux em seu laptop. Em menos de 5 minutos aparecer? algu?m pra discordar de sua escolha de distribui??o, do particionamento, do gerenciador de janelas, do ambiente de desktop, do editor de textos..." -------------- next part -------------- An HTML attachment was scrubbed... URL: From otte at gnome.org Wed Sep 30 16:06:04 2009 From: otte at gnome.org (Benjamin Otte) Date: Wed, 30 Sep 2009 16:06:04 +0200 Subject: [gst-devel] Using cairo/pixman for raw video in GStreamer In-Reply-To: References: Message-ID: So, here is an update on what happened on this in the last 3 weeks. (Warning: It might get quite in depth in both Cairo and GStreamer terminology, so if you don't know about some things I'm talking about, don't hesitate to ask me about it in a reply or on IRC. I'll probably assume more than rudimentary GStreamer and Cairo knowledge in this mail. I want to keep it short and to the point.) When I'm talking about test results, those were created on a Macbook 2.2 with an Intel 945 GPU on Karmic. Don't expect this to be as performant on old hard-/software. I would however expect it to be as performant on recent X servers with Intel and Radeons. But you have been warned. :) What have I done so far? I've written code to implement my ideas. The code exists in public git branches and is expected to work, should you want to test it. The code should compile fine on any somewhat recent distro. Read: If you can compile git master of gstreamer and cairo, you can compile this code. Of course, it's alpha quality, so expect it to change quickly. But it should definitely compile and run. pixman: http://cgit.freedesktop.org/~company/pixman/log/?h=yuv I added support for most YUV formats that GStreamer supports today. The missing ones weren't added because they weren't necessary to prove my point. I also enhanced the API to allow creating planar images. The code is not yet optimized in any way, but I intend to hook in David's ORC code to accelerate common YUV operations. cairo: http://cgit.freedesktop.org/~company/cairo/log/?h=yuv I exported one function to be able to use any pixman format and be able to create planar image surfaces. A bunch of bugfixes were necessary, too. They're all landed in git master though. gst-plugins-cairo: http://cgit.freedesktop.org/~company/gst-plugins-cairo This new repository contains a library libgstcairo and a bunch of plugins using that library. The library does three things. First it abstracts the caps handling. This allows adding new caps to the library without the need to update the elements. It also adds a bunch of support functions that make writing caps nego code a lot simpler. Second, it contains code to create cairo surfaces from GstBuffers and vice versa. And last but not least it introduces a new format "video/x-cairo" that allows passing cairo surfaces in buffers. As this is all done transparently, elements will render to GL or whatever surfaces the moment they become available to libgstcairo without the need to recompile them. The elements implement the functionality of the most common GStreamer raw video elements. So far, there are (in order of creation and with the elements they're intended to replace): - cairocolorspace (ffmpegcolorspace) - puzzle - cairotestsrc (videotestsrc) - pangotimeoverlay (timeoverlay) - cairoxsink (ximagesink/xvmagesink) - cairoscale (videoscale) I'll code more elements and implement features for the current ones as I get around to it. In particular a full videomixer and textoverlay replacement are on my list. These elements are parrallel-installable to curent GStreamer elements; they will not override any existing elements. What did I learn so far? Cairo looks like the perfect match for GStreamer video handling, even when talking about memory buffers only. I was surprised at how quickly I could achieve progress and that there are no features I had to leave behind while porting elements. In fact, elements gained features because they support more video/x-raw-* formats now than they did before. Also, the code required got a LOT smaller. Most current elements duplicate the code to handle formats (wc -l for elements: ffmpegcolorspace: 7400, videoscale: 4600, videotestsrc: 3250) while gst-cairo hooks into the Cairo code with little overhead (libgstcairo: 1100 lines, cairocolorspace: 150, cairotestsrc: 900, cairoscale: 170). So we can talk about orders of magnitude of code that gets saved while not losing features. The performance when compared wth the default elements is somewhere between 5x slower and 3x faster, depending on what one is doing and without me having done any optimizations. I expect performance to be at least equal to current code but likely better once all optimizations are hooked up. So much for backwards compatibility. (So much for Cairo being "slow", when it can beat GStreamer noticably in some cases. ;)) The interesting thing is video/x-cairo. This can allow running a whole pipeline on the GPU without any need to move the data in main memory. When it works, its performance improvements can be measured in orders of magnitude again. A simple example (real/user times from "time", gst-launch pipeline used): 8.631s - 4.712s - videotestsrc num-buffers=1000 ! video/x-raw-yuv,width=800,height=600 ! xvimagesink sync=false 6.581s - 4.564s - videotestsrc num-buffers=1000 ! video/x-raw-rgb,width=800,height=600 ! ximagesink sync=false 0.632s - 0.488s - cairotestsrc num-buffers=1000 ! video/x-cairo,width=800,height=600 ! cairoxsink sync=false Or a somewhat more demanding example: 18.843s - 15.585s - videotestsrc num-buffers=1000 ! timeoverlay ! video/x-raw-yuv,width=800,height=600 ! xvimagesink sync=false 21.552s - 17.237s - videotestsrc num-buffers=1000 ! timeoverlay ! video/x-raw-rgb,width=800,height=600 ! ximagesink sync=false 1.187s - 0.668s - cairotestsrc num-buffers=1000 ! pangotimeoverlay ! video/x-cairo,width=800,height=600 ! cairoxsink sync=false Getting these performance gains requires access to hardware buffers in the whole pipeline. And the current design of hardware access libraries (both GL and X) and GStreamer doesn't make it any easier. Which brings us to the next point: What are the remaining issues? First of all: For memory buffers there are no remaining issues. You can probably use cairocolorspace and cairoscale as drop-in replacements without any issues today. With that said, the one big remaining issue is: Get things reliably hardware-accelerated. There's a noticable difference between all elements being accelerated and all but one element being accelerated. While the code falls back to software rendering whenever something is not supported - so there's no internal flow errors or even crashes - but it's a performance difference. Usually it's the difference between no CPU usage and one busy core. (on a lighter note, with a CPU meter it's easy to detect if the whole pipeline is properly accelerated.) Here's a list of isues I'm facing (from higher to lower layer): - GStreamer threading Code that involves GstBuffers can be called pretty much by any thread at any time - to be exact: buffers can be read by multiple threads, but only one thread at a time may write to it. This thread however may change. So there is a challenge in making sure that cairo surfaces that are kept inside buffers don't step on each other's toes from multiple threads. This is not an issue with image surfaces, but it is an issue with at least GL and X. Not sure about DirectFB, DirectX or DRM, but I'd suspect they have similar issues. Getting this right is possible today, but it would be nice if GStreamer would tell buffers when they are passing a thread boundary. I suspect this is not easy to do before 0.11 though. - GStreamer buffer allocation Buffer allocation code has quite some places where it simply returns a memory buffer when caps do not match. While gstcairo handles this fine and falls back to software rendering, it'll get slow. So making sure that in a pipeline like cairotestsrc ! video/x-cairo,width=800,height=600 ! cairoscale ! video/x-cairo,width=600,height=450 ! cairoxsink the cairotestsrc can allocate a 800x600 buffer from cairoxsink is desirable. While some ideas exist on how to make this work, there's no supported way of making it happen. - Cairo meta surfaces copy whole surfaces One way of solving the aforementioned issue and a nice way to handle the threading issues listed above is to use meta surfaces and only replay them in the sink element. Unfortunately, Cairo copies image buffers when generating snapshots, which kills performance right there. It would be nice if there was a way to keep surfaces unchanged until they are modified (copy in begin_modification maybe). If I comment out that code, this solution works fast today. It has one drawback though: As meta surfaces keep references to source surfaces around, we'd need to have threadsafe source surfaces. So we'd either need to not use GL/X surfaces at all (and rely on meta surfaces only) or find a way to copy the surfaces when moving over thread boundaries. - Rendering to subsampled images I didn't spend a lot of time after finding an ok solution, but it's a challenging task to support rendering to vertically subsampled images with pixman's scanline based approach. On one hand this is a pretty futile attempt, as the subsampling will result in artifacts no matter what one does, but on the other hand it'd be nice if rendering would work, so one could support subtitles or even overlays as seen on TV. And the most-used YUV format (I420/YV12) is horizontally subsampled. It's not terribly important as in most cases conversion can be done as the last step, but if somebody has a solution to the problem, I'm all ears. - Ways to upload YUV data There is currently no good way to get an accelerated upload of YUV data to X. The only ways I'm aware of are GL (see gst-plugins-gl for code) and xv. So either we'll need to continue converting to RGB in the X case and focus on using cairo-gl, make cairo use xv or convince the X people that such a thing as YUV uploads would be a welcome addition to Xrender. It seems X people are currently at XDC getting all excited about Wayland, so I don't have very high hopes. - Using the GPU's video decoding abilities As GPUs can decode videos in all the recent formats, it would be nice if there could be elements that take raw MPEG, H264 or whatever frames and stuck the result into a cairo buffer, preferrably in the GPU. This would also get around the issue above for most videos people watch today. Unfortunately it seems the people involved in these projects haven't yet figured out if they want to name it vaapi, vdpau, xvba or xvmc (alphabetical order here, I have no preferences), what capitalization scheme they want to use or if they even want to make it open source. So it seems this is going nowhere in the forseeable future, too. What now? When talking on IRC about this I realized that there is quite a lack of knowledge on all sides - my knowledge often isn't deep enough, GStreamer people don't know enough about state-of-the-art video handling and its gains and pitfalls, Cairo and X people lack knowledge about the requirements for video playback - and this lack of knowledge often results in preconceptions that lead to wrong decisions and make life harder on all sides. It would be nice if there was a way to get you people together and actually educate each other about this process. I'd suggest a hackfest, but I'm not sure what others think and who to approach for funding and locations. There's also the issue of the maintainers' opinions about this code. As there is quite a few projects involved (GStreamer, Cairo, pixman, and possibly X) and I'm not really interested in spending a lot of work on polishing code that ends up in some demo repository or gets rejected. It'd also be nice to get a review rather sooner than later so I can fix design issues that are in need of updating while not having so much code depend on it. And of course, I like people reviewing and complimenting me on my code. :) Then there's gst-plugins-gl. The GL plugins and my work touch on some of the same issues (most notably hardware acceleration) and I'd like to make sure this code can work with their approach and make use cairo-gl buffers internally. The best possible outcome from my point of view would be if we could port the GL plugins to use gstcairo and make gstcairo provide the required functionality. That way we'd get rid of the need to put explicit upload and download elements and gain the ability to do all the GL stuff. Unfortunately I lack knowledge about GL, so it'd be nice if someone else could look at that. I think that's all for now, Benjamin From dick at mrns.nl Wed Sep 30 18:17:34 2009 From: dick at mrns.nl (Dick Marinus) Date: Wed, 30 Sep 2009 18:17:34 +0200 Subject: [gst-devel] using the tuner interface from python Message-ID: <1254327454.4470.8.camel@latitude> Hi all, At April 30, 2006 Fredrik posted a message about using the tuner interface from Python and I'm experiencing the same issue: $ python >>> import gst >>> source = gst.element_factory_make("v4l2src",name="Pinnacle PCTV USB2.0") >>> print source.get_channel() (.:13363): GStreamer-CRITICAL **: gst_implements_interface_cast: assertion `gst_element_implements_interface (GST_ELEMENT (from), iface_type)' failed ** (.:13363): CRITICAL **: gst_tuner_get_channel: assertion `GST_IS_TUNER (tuner)' failed I've tried to set the pipeline queue to STATE_PLAYING but I still have the same error. I guess something is broken at the Python bindings. Could someone please help me? I'm using: gst-python=0.10.16 gstreamer=0.10.24 gst-plugins-good=0.10.16 gst-plugins-base=0.10.24 Thanks in advance, Dick From gerecke at gmail.com Tue Sep 15 22:18:26 2009 From: gerecke at gmail.com (attaboy) Date: Tue, 15 Sep 2009 13:18:26 -0700 (PDT) Subject: [gst-devel] UDPSRC BUG In-Reply-To: <3c1737210909150959y2d51346bibf060169902dc9c@mail.gmail.com> References: <881487A9950B044AA6861B700374AAAF02C3C012@deimsg40.de.net.world> <180a127d0909150852mca47c9ela0b1325589aa512b@mail.gmail.com> <3c1737210909150959y2d51346bibf060169902dc9c@mail.gmail.com> Message-ID: <25461052.post@talk.nabble.com> I can confirm that udpsrc is broken on Windows and I'm willing to test fixes. I've just spent more time than I'd like to admit trying to get a multi-cast receive working on Windows XP using this simple pipeline: gst-launch udpsrc uri="udp://224.1.2.3:1234" ! fakesink I tried the 0.10.15 udpsrc plugin with this patch (https://forja.rediris.es/forum/message.php?msg_id=180551) and had no success with failure on the bind. I backed up to 0.10.8 (before IP6 support) and still had failure on the bind. I hard coded the local address to INADDR_ANY right before the bind and success! /* if (src->multi_addr.imr_multiaddr.s_addr) src->myaddr.sin_addr.s_addr = src->multi_addr.imr_multiaddr.s_addr; else */ src->myaddr.sin_addr.s_addr = INADDR_ANY; I tried hardwiring INADDR_ANY with the patched 0.10.15 version and ran with no crashes, but also received no data, so right now I'm back to 0.10.8. Point me to some code/patches and I'll test on W2K3, XP and Windows 7. I'm building using the OSS build system as opposed to mingw, hopefully that's not a problem. Bill Michael Smith-59 wrote: > > On Tue, Sep 15, 2009 at 8:52 AM, Julien Isorce > wrote: >> Hi, >> >> Does this pipeline works: gst-launch-0.10.exe udpsrc port=200 ! fakesink >> ? >> >> If not, try to compile the udp plugin with mingw >> >>>>Then I build the newest Trunk Version form the SVN. The result is the >>>> same. >> >> if you are currently using VS. >> >> (with the current git, udpsink fails with this error (and compiled with >> mingw) >> WARN????????? multiudpsink >> gstmultiudpsink.c:790:gst_multiudpsink_init_send: error: Could >> not >> set TTL socket option (0): No error > > I believe this patch makes multiudpsink work on windows: > https://bugzilla.gnome.org/show_bug.cgi?id=534243 > > Note that I'm still waiting for more windows testing (particularly on > vista), and hopefully a code review from someone who knows this stuff. > > udpsrc might also have similar issues, I'm not sure - I can fix these > if people are willing to test the code and comment on that bug. > > Mike > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register > now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > -- View this message in context: http://www.nabble.com/UDPSRC-BUG-tp25455747p25461052.html Sent from the GStreamer-devel mailing list archive at Nabble.com. From balakrishnan.2005 at gmail.com Wed Sep 16 11:08:17 2009 From: balakrishnan.2005 at gmail.com (Balakrishnan Natarajan) Date: Wed, 16 Sep 2009 14:38:17 +0530 Subject: [gst-devel] what is pipeline for avi to mp4 transcoding Message-ID: <823065aa0909160208u39b43d91scd44a75f3ed1a6d6@mail.gmail.com> Hi All, I am looking for a Gstreamer pipeline to do the following. - Convert from AVI format to MP4 Format, *The MP4 should contain the same the elementary Audio and Video streams as that of AVI* * * test_h264_aac.avi : (Video.h264, Audio.aac) -----------> output_h264_aac.mp4 : (Video.h264, Audio.aac) *So I do not want to decode the elementary streams and again encode it back before mux-ing with the MP4* I tried with the below pipeline. * gst-launch-0.10 filesrc location=test_h264_aac.avi ! avidemux name=demux { qtmux name=mux ! filesink location=output.mp4 } { demux. ! queue ! audiopass ! mux. } { demux. ! queue ! videopass ! mux. }* Here *audiopass, and **videopass *are gstreamer plugins, which pass through the audio and video elementary streams as it. These elements are used only for caps negotiation of the audio and video src pads of AVI demuxer with the audio and video sink pads of the MP4 muxer. This pipeline does not write out any Data. Can somebody please suggest me a pipeline to do the conversion mentioned above. Thanks and Regards Bala -------------- next part -------------- An HTML attachment was scrubbed... URL: From balakrishnan.2005 at gmail.com Thu Sep 17 14:14:21 2009 From: balakrishnan.2005 at gmail.com (Balakrishnan Natarajan) Date: Thu, 17 Sep 2009 17:44:21 +0530 Subject: [gst-devel] Gstreamer pipeline to convert from avi to mp4 format Message-ID: <823065aa0909170514k7dce2ecet2b34bb4075c530e5@mail.gmail.com> > > Hi Aurelin and All > Thanks for your reply. I am replying on behalf of Pawan The output of gst-launch is as below. balakrishnan at bglldtw126:~/Bala/Work$ gst-launch-0.10 filesrc location=test_h264_aac.avi ! avidemux name=demux { qtmux name=mux ! filesink location=output.mp4 } { demux. ! queue ! audiopass ! mux. } { demux. ! queue ! videopass ! mux. } (gst-launch-0.10:18379): GLib-GObject-WARNING **: cannot register existing type `GstSignalProcessor' (gst-launch-0.10:18379): GLib-CRITICAL **: g_once_init_leave: assertion `initialization_value != 0' failed (gst-launch-0.10:18379): GLib-GObject-WARNING **: cannot retrieve class for invalid (unclassed) type `' Setting pipeline to PAUSED ... Pipeline is PREROLLING ... Setting CapsSetting CapsPipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstSystemClock So the pipeline goes into playing mode and is stuck in there. So at this instant the output.mp4 is 0 bytes in size However when i forcefully (ctrl C) terminate the pipeline the output.mp4 ( find attached with the mail) is 317 bytes in size which is of course not playable. I have attached the logs with the following option GST_DEBUG=qtmux:5,filesink:5,queue:5 Looking att that i feel the pipeline is somewhere stuck in the queue. the avidmux continuously runs with out any problems so have not taken the same. I have also attached the sources for audiopass and the videopass elements mentioned below. Please see the same and do let me know your observations Thanks Bala > > > > ------------------------------ > > Message: 3 > Date: Thu, 17 Sep 2009 09:27:59 +0200 > From: Aurelien Grimaud > Subject: Re: [gst-devel] Gstreamer pipeline to convert from avi to mp4 > format !! > To: Discussion of the development of GStreamer > > Message-ID: <4AB1E4FF.3070909 at yahoo.fr> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > > Could you post the output of your gst-launch ? What does "does not write > out any data" means ? Stuck in preroll ?, not negotiated ? > Did you try to GST_DEBUG it ? > Should give hints on what happens ... > > Aurelien > > Kumar, Pawan a ?crit : > > > > Hi All, > > > > I am looking for a Gstreamer pipeline to convert from avi to mp4 > > format. Here mp4 should contain the same the elementary Audio and > > Video streams as stored in avi. > > > > test_h264_aac.avi : (Video.h264, Audio.aac) -----------> > > output_h264_aac.mp4 : (Video.h264, Audio.aac) > > > > So I do not want to decode the elementary streams and again encode > > it back before mux-ing with the MP4 > > > > I tried with the below pipeline. > > > > gst-launch-0.10 filesrc location=test_h264_aac.avi ! avidemux > > name=demux { qtmux name=mux ! filesink location=output.mp4 } { demux. > > ! queue ! audiopass ! mux. } { demux. ! queue ! videopass ! mux. } > > > > Here audiopass, and videopass are gstreamer plugins, which pass > > through the audio and video elementary streams as it. These elements > > are used only for caps negotiation of the audio and video src pads of > > AVI demuxer with the audio and video sink pads of the MP4 muxer. > > > > This pipeline does not write out any Data. > > > > Can somebody please suggest me a pipeline to do the conversion > > mentioned above ? > > > > > > Thanks, > > /Pawan > > > > ------------------------------------------------------------------------ > > > > > ------------------------------------------------------------------------------ > > Come build with us! The BlackBerry® Developer Conference in SF, CA > > is the only developer event you need to attend this year. Jumpstart your > > developing skills, take BlackBerry mobile applications to market and stay > > ahead of the curve. Join us from November 9-12, 2009. Register > now! > > http://p.sf.net/sfu/devconf > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > gstreamer-devel mailing list > > gstreamer-devel at lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > > -------------- next part -------------- > An HTML attachment was scrubbed... > > -- - bala -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- gst-launch-0.10 filesrc location=test_h264_aac.avi ! avidemux name=demux { qtmux name=mux ! filesink location=output.mp4 } { demux. ! queue ! audiopass ! mux. } { demux. ! queue ! videopass ! mux. } (gst-launch-0.10:18261): GLib-GObject-WARNING **: cannot register existing type `GstSignalProcessor' (gst-launch-0.10:18261): GLib-CRITICAL **: g_once_init_leave: assertion `initialization_value != 0' failed (gst-launch-0.10:18261): GLib-GObject-WARNING **: cannot retrieve class for invalid (unclassed) type `' 0:00:02.199724219 18260 0x8cf5050 LOG qtmux gstqtmux.c:1837:gst_qt_mux_register: Registering muxers 0:00:02.200871037 18260 0x8cf5050 LOG qtmux gstqtmux.c:1866:gst_qt_mux_register: Finished registering muxers 0:00:02.202227248 18260 0x8cf5050 DEBUG queue gstqueue.c:419:gst_queue_init: initialized queue's not_empty & not_full conditions 0:00:02.203201307 18260 0x8cf5050 DEBUG queue gstqueue.c:419:gst_queue_init: initialized queue's not_empty & not_full conditions 0:00:02.204413668 18260 0x8cf5050 DEBUG qtmux gstqtmux.c:1645:gst_qt_mux_request_new_pad: Requested pad: (NULL) 0:00:02.204753868 18260 0x8cf5050 DEBUG queue gstqueue.c:489:gst_queue_link_src: queue linking source pad 0:00:02.204777576 18260 0x8cf5050 DEBUG queue gstqueue.c:502:gst_queue_link_src: not starting task reason wrong-state 0:00:02.204987905 18260 0x8cf5050 DEBUG qtmux gstqtmux.c:1645:gst_qt_mux_request_new_pad: Requested pad: (NULL) 0:00:02.205322587 18260 0x8cf5050 DEBUG queue gstqueue.c:489:gst_queue_link_src: queue linking source pad 0:00:02.205344701 18260 0x8cf5050 DEBUG queue gstqueue.c:502:gst_queue_link_src: not starting task reason wrong-state Setting pipeline to PAUSED ... 0:00:02.206819575 18260 0x8cf5050 DEBUG filesink gstfilesink.c:434:gst_file_sink_do_seek: Seeking to offset 0 using fseeko 0:00:02.206859738 18260 0x8cf5050 DEBUG filesink gstfilesink.c:345:gst_file_sink_open_file: opened file output.mp4, seekable 1 Pipeline is PREROLLING ... 0:00:02.247369636 18260 0x8e230a8 DEBUG queue gstqueue.c:577:apply_segment: configured NEWSEGMENT time segment start=0:00:00.000000000, stop=4:38:46.401451247, last_stop=0:00:00.000000000, duration=99:99:99.999999999, rate=1.000000, applied_rate=1.000000, flags=0x00, time=0:00:00.000000000, accum=0:00:00.000000000 0:00:02.247461685 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.000000000, src 0:00:00.000000000 0:00:02.247504960 18260 0x8dfab30 DEBUG queue gstqueue.c:577:apply_segment: configured NEWSEGMENT time segment start=0:00:00.000000000, stop=4:38:46.401451247, last_stop=0:00:00.000000000, duration=99:99:99.999999999, rate=1.000000, applied_rate=1.000000, flags=0x00, time=0:00:00.000000000, accum=0:00:00.000000000 0:00:02.247548133 18260 0x8dfab30 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.000000000, src 0:00:00.000000000 0:00:02.247594708 18260 0x8e230a8 DEBUG queue gstqueue.c:577:apply_segment: configured NEWSEGMENT time segment start=0:00:00.000000000, stop=4:38:46.401451247, last_stop=0:00:00.000000000, duration=99:99:99.999999999, rate=1.000000, applied_rate=1.000000, flags=0x00, time=0:00:00.000000000, accum=0:00:00.000000000 0:00:02.247625298 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.000000000, src 0:00:00.000000000 0:00:02.247654023 18260 0x8e25160 DEBUG queue gstqueue.c:577:apply_segment: configured NEWSEGMENT time segment start=0:00:00.000000000, stop=4:38:46.401451247, last_stop=0:00:00.000000000, duration=99:99:99.999999999, rate=1.000000, applied_rate=1.000000, flags=0x00, time=0:00:00.000000000, accum=0:00:00.000000000 0:00:02.247695957 18260 0x8e25160 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.000000000, src 0:00:00.000000000 0:00:02.247754694 18260 0x8dfab30 DEBUG qtmux gstqtmux.c:1610:gst_qt_mux_sink_event: received tag event 0:00:02.247839154 18260 0x8e25160 DEBUG qtmux gstqtmux.c:1610:gst_qt_mux_sink_event: received tag event 0:00:02.247904730 18260 0x8e25160 DEBUG qtmux gstqtmux.c:1610:gst_qt_mux_sink_event: received tag event 0:00:02.247979038 18260 0x8dfab30 DEBUG qtmux gstqtmux.c:1610:gst_qt_mux_sink_event: received tag event 0:00:02.248162693 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.040000000 0:00:02.248184392 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.040000000, src 0:00:00.000000000 0:00:02.248209679 18260 0x8dfab30 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.040000000 0:00:02.248224777 18260 0x8dfab30 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.040000000, src 0:00:00.040000000 Setting Caps0:00:02.248407256 18260 0x8dfab30 DEBUG qtmux gstqtmux.c:1411:gst_qt_mux_video_sink_set_caps: mux:pad0, caps=video/x-h264, variant=(string)itu, framerate=(fraction)25/1, width=(int)384, height=(int)288, pixel-aspect-ratio=(fraction)1/1 0:00:02.248452171 18260 0x8dfab30 DEBUG qtmux gstqtmux.c:1446:gst_qt_mux_video_sink_set_caps: Rate of video track selected: 2500 0:00:02.248466366 18260 0x8dfab30 WARN qtmux gstqtmux.c:1507:gst_qt_mux_video_sink_set_caps: no codec_data in h264 caps 0:00:02.248588752 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.023219954 0:00:02.248604920 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.023219954, src 0:00:00.000000000 0:00:02.248628899 18260 0x8e25160 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.023219954 0:00:02.248642569 18260 0x8e25160 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.023219954, src 0:00:00.023219954 Setting Caps0:00:02.248788533 18260 0x8e25160 DEBUG qtmux gstqtmux.c:1186:gst_qt_mux_audio_sink_set_caps: mux:pad1, caps=audio/mpeg, mpegversion=(int)4, rate=(int)44100, channels=(int)2 0:00:02.248815640 18260 0x8e25160 WARN qtmux gstqtmux.c:1247:gst_qt_mux_audio_sink_set_caps: no (valid) codec_data for AAC audio 0:00:02.248850542 18260 0x8e25160 DEBUG qtmux gstqtmux.c:871:gst_qt_mux_start_file: starting file 0:00:02.248882579 18260 0x8e25160 DEBUG qtmux gstqtmux.c:841:gst_qt_mux_send_ftyp: Sending ftyp atom 0:00:02.248906690 18260 0x8e25160 LOG qtmux gstqtmux.c:850:gst_qt_mux_send_ftyp: Pushing ftyp 0:00:02.248918675 18260 0x8e25160 LOG qtmux gstqtmux.c:564:gst_qt_mux_send_buffer: sending buffer size 20 0:00:02.248929626 18260 0x8e25160 LOG qtmux gstqtmux.c:577:gst_qt_mux_send_buffer: downstream 0:00:02.249017500 18260 0x8e25160 DEBUG filesink gstfilesink.c:501:gst_file_sink_event: Ignored NEWSEGMENT, no seek needed Pipeline is PREROLLED ... 0:00:02.249055981 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:01.671836734 0:00:02.249088655 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:01.671836734, src 0:00:00.023219954 Setting pipeline to PLAYING ... 0:00:02.249123566 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.080000000 0:00:02.249139885 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.080000000, src 0:00:00.040000000 0:00:02.249181897 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.120000000 0:00:02.249195326 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.120000000, src 0:00:00.040000000 0:00:02.249229750 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.160000000 0:00:02.249242671 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.160000000, src 0:00:00.040000000 0:00:02.249282499 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.200000000 0:00:02.249295363 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.200000000, src 0:00:00.040000000 0:00:02.249329642 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.240000000 0:00:02.249352482 18260 0x8e25160 DEBUG filesink gstfilesink.c:573:gst_file_sink_render: writing 20 bytes at 0 0:00:02.254613206 18260 0x8e25160 DEBUG qtmux gstqtmux.c:672:gst_qt_mux_send_mdat_header: Sending mdat's atom header, size 0 0:00:02.254635856 18260 0x8e25160 LOG qtmux gstqtmux.c:692:gst_qt_mux_send_mdat_header: Pushing mdat start 0:00:02.254647631 18260 0x8e25160 LOG qtmux gstqtmux.c:564:gst_qt_mux_send_buffer: sending buffer size 16 0:00:02.254658328 18260 0x8e25160 LOG qtmux gstqtmux.c:577:gst_qt_mux_send_buffer: downstream 0:00:02.254674993 18260 0x8e25160 DEBUG filesink gstfilesink.c:573:gst_file_sink_render: writing 16 bytes at 20 0:00:02.254691192 18260 0x8e25160 LOG qtmux gstqtmux.c:1146:gst_qt_mux_collected: selected pad pad1 with time 0:00:00.000000000 0:00:02.254708115 18260 0x8e25160 LOG qtmux gstqtmux.c:948:gst_qt_mux_add_buffer: Pad pad1 has no previous buffer stored, storing now 0:00:02.254723869 18260 0x8e25160 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:01.671836734 0:00:02.254737117 18260 0x8e25160 LOG queue gstqueue.c:541:update_time_level: sink 0:00:01.671836734, src 0:00:01.671836734 0:00:02.254755739 18260 0x8e25160 LOG qtmux gstqtmux.c:1146:gst_qt_mux_collected: selected pad pad0 with time 0:00:00.000000000 0:00:02.254768281 18260 0x8e25160 LOG qtmux gstqtmux.c:948:gst_qt_mux_add_buffer: Pad pad0 has no previous buffer stored, storing now 0:00:02.249365775 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.240000000, src 0:00:00.040000000 0:00:02.254795427 18260 0x8dfab30 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.080000000 0:00:02.254808958 18260 0x8dfab30 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.240000000, src 0:00:00.080000000 0:00:02.254826937 18260 0x8dfab30 LOG qtmux gstqtmux.c:1146:gst_qt_mux_collected: selected pad pad1 with time 0:00:00.023219954 0:00:02.254844239 18260 0x8dfab30 LOG qtmux gstqtmux.c:1023:gst_qt_mux_add_buffer: Pad (pad1) dts updated to 0:00:00.023219954 0:00:02.254856167 18260 0x8dfab30 LOG qtmux gstqtmux.c:1027:gst_qt_mux_add_buffer: Adding 1 samples to track, duration: 1023 size: 281 chunk offset: 0 0:00:02.254870621 18260 0x8dfab30 LOG qtmux gstqtmux.c:564:gst_qt_mux_send_buffer: sending buffer size 281 0:00:02.254881391 18260 0x8dfab30 LOG qtmux gstqtmux.c:577:gst_qt_mux_send_buffer: downstream 0:00:02.254898242 18260 0x8dfab30 DEBUG filesink gstfilesink.c:573:gst_file_sink_render: writing 281 bytes at 36 0:00:02.254947406 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.280000000 0:00:02.254961259 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.280000000, src 0:00:00.080000000 New clock: GstSystemClock 0:00:02.266750353 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.320000000 0:00:02.266769673 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.320000000, src 0:00:00.080000000 0:00:02.266807072 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.360000000 0:00:02.266820759 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.360000000, src 0:00:00.080000000 0:00:02.266860582 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.400000000 0:00:02.266873683 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.400000000, src 0:00:00.080000000 0:00:02.266906880 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.440000000 0:00:02.266919833 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.440000000, src 0:00:00.080000000 0:00:02.266951082 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.480000000 0:00:02.266964065 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.480000000, src 0:00:00.080000000 0:00:02.266996986 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.520000000 0:00:02.267009849 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.520000000, src 0:00:00.080000000 0:00:02.267039798 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.560000000 0:00:02.267052673 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.560000000, src 0:00:00.080000000 0:00:02.267081750 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.600000000 0:00:02.267094574 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.600000000, src 0:00:00.080000000 0:00:02.267127798 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.640000000 0:00:02.267140637 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.640000000, src 0:00:00.080000000 0:00:02.267170355 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.680000000 0:00:02.267183323 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.680000000, src 0:00:00.080000000 0:00:02.267213591 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.720000000 0:00:02.267226433 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.720000000, src 0:00:00.080000000 0:00:02.267263766 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.760000000 0:00:02.267276957 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.760000000, src 0:00:00.080000000 0:00:02.283896484 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.800000000 0:00:02.283911215 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.800000000, src 0:00:00.080000000 0:00:02.283941489 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.840000000 0:00:02.283954542 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.840000000, src 0:00:00.080000000 0:00:02.283988791 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.880000000 0:00:02.284001940 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.880000000, src 0:00:00.080000000 0:00:02.284034786 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.920000000 0:00:02.284047790 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.920000000, src 0:00:00.080000000 0:00:02.284080223 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:00.960000000 0:00:02.284093162 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:00.960000000, src 0:00:00.080000000 0:00:02.284123424 18260 0x8e230a8 LOG queue gstqueue.c:603:apply_buffer: last_stop updated to 0:00:01.000000000 0:00:02.284136482 18260 0x8e230a8 LOG queue gstqueue.c:541:update_time_level: sink 0:00:01.000000000, src 0:00:00.080000000 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Name: audiopass.c Type: text/x-csrc Size: 9319 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: videopass.c Type: text/x-csrc Size: 8325 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: videopass.h Type: text/x-chdr Size: 3237 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: output.mp4 Type: video/mp4 Size: 317 bytes Desc: not available URL: From cworth at cworth.org Wed Sep 30 18:48:55 2009 From: cworth at cworth.org (Carl Worth) Date: Wed, 30 Sep 2009 09:48:55 -0700 Subject: [gst-devel] [cairo] Using cairo/pixman for raw video in GStreamer In-Reply-To: References: Message-ID: <1254328466-sup-8795@yoom.home.cworth.org> Excerpts from Benjamin Otte's message of Wed Sep 30 07:06:04 -0700 2009: > What have I done so far? Fantastic stuff, Benjamin! Thanks for doing all this and writing it all up. > - Cairo meta surfaces copy whole surfaces > One way of solving the aforementioned issue and a nice way to handle > the threading issues listed above is to use meta surfaces and only > replay them in the sink element. Unfortunately, Cairo copies image > buffers when generating snapshots, which kills performance right > there. The meta-surface image-copying predates a bunch of copy-on-write work in cairo for more efficient snapshots. It should be the case that what the meta-surface does here is to create a *snapshot* of each surface, (not an explicit copy). And if that's the case, and it's simply a matter of making the snapshot code just delay that copy until actually needed, then that should be easy with the infrastructure in place in cairo now. > - Ways to upload YUV data > There is currently no good way to get an accelerated upload of YUV > data to X. The only ways I'm aware of are GL (see gst-plugins-gl for > code) and xv. So either we'll need to continue converting to RGB in > the X case and focus on using cairo-gl, make cairo use xv or convince > the X people that such a thing as YUV uploads would be a welcome > addition to Xrender. > It seems X people are currently at XDC getting all excited about > Wayland, so I don't have very high hopes. I don't understand your final point here at all. Having a bunch of X people together at the same time seems an ideal way to make progress here. I'll go ahead and read your paragraph here to the group this morning and let you know what comes of it. > When talking on IRC about this I realized that there is quite a lack > of knowledge on all sides - my knowledge often isn't deep enough, > GStreamer people don't know enough about state-of-the-art video > handling and its gains and pitfalls, Cairo and X people lack knowledge > about the requirements for video playback - and this lack of knowledge > often results in preconceptions that lead to wrong decisions and make > life harder on all sides. > It would be nice if there was a way to get you people together and > actually educate each other about this process. I'd suggest a > hackfest, but I'm not sure what others think and who to approach for > funding and locations. The Linux Plumbers Conference was conceived of precisely to be able to address these kinds of cross-project issues that are hard to solve when many get-togethers address each project in isolation. Unfortunately, Plumbers just happened last week so the next one will be a year away. As for funding, the X.org Foundation has funds and is very interested in supporting X-related development events like this. The X.org Foundation Board of Directors just stood up in XDC and asked for people to request funds they need to get work done, (whether for hardware or for hackfests, etc.). So I think this is an ideal case. When you have some details put together, please feel free to email the board as a whole (xf_board at x.org) or me individually, (as a board member for at least the next few months). Again, great work! -Carl -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 190 bytes Desc: not available URL: From rohan at perzonae.com Wed Sep 23 12:05:09 2009 From: rohan at perzonae.com (Rohan) Date: Wed, 23 Sep 2009 12:05:09 +0200 Subject: [gst-devel] Is it possible to stream video and audio with udp on separate streams? Message-ID: <4AB9F2D5.7070509@perzonae.com> Hi all, I am quite new to gstreamer, and have been messing around with streaming audio and video. It is important for what I am doing that the audio and video streams are kept separate. The video is also output on the client side. I have got most of this working with these two scripts: --------------------------------------------------------- #!/bin/bash #### Image/sound source, sends signal to receiver # tap the v4l2src and tee it pipe='v4l2src ! tee name=tee ' # original video source, and fork # fork off, buffer pipe="$pipe tee. ! queue ! ffmpegcolorspace ! smokeenc keyframe=8 qmax=40 ! " # send to server pipe="$pipe udpsink host=127.0.0.1 port=5000 sync=false " # fork off, buffer, and display pipe="$pipe tee. ! queue ! " # encode and display pipe="$pipe ffmpegcolorspace ! xvimagesink" # and now to add sound pipe="$pipe audiotestsrc ! queue ! decodebin ! " pipe="$pipe audioconvert ! speexenc ! tcpserversink host=127.0.0.1 port=5001 " gst-launch-0.10 $pipe --------------------------------------------------------- #!/bin/bash # video receiver and display # video src pipe="udpsrc port=5000 ! smokedec ! autovideosink" # audio pipe="$pipe tcpclientsrc host=127.0.0.1 port=5001 ! " pipe="$pipe speexdec ! queue ! alsasink sync=false " gst-launch-0.10 $pipe ----------------------------------------------------------- What I would prefer to do is send both streams over udp, but when I change the tcp parts to the same as used for video, but with a different port, e.g. 5001 the audio does not work. Is there something I am missing, or is it not possible to stream two separate udp streams in this way? Thanks for any help, Rohan P.S. I have also been having problems getting video bins to play nicely, but that is for another post. ;) From rohan at perzonae.com Wed Sep 23 13:47:12 2009 From: rohan at perzonae.com (Rohan) Date: Wed, 23 Sep 2009 13:47:12 +0200 Subject: [gst-devel] creating bins and ghostpads Message-ID: <4ABA0AC0.7090909@perzonae.com> Hi all, I am obviously doing something wrong here, but cannot figure out what. Essentially the pipeline is the same as the receiver but I have broken it into bins. I seem to be having problems with the ghostpad, because I get this error: (stream_receiver.py:1914): GStreamer-WARNING **: Trying to connect elements that don't share a common ancestor: vidsource and vidbin Traceback (most recent call last): File "stream_receiver.py", line 118, in server() File "stream_receiver.py", line 27, in __init__ vidsource.link(vidbin) gst.LinkError: failed to link vidsource with vidbin Here is the code: ---------------------------------------------------------------------- #!/bin/env python # The gst pipeline # pipe="udpsrc port=5000 ! smokedec ! autovideosink" # # audio # pipe="$pipe tcpclientsrc host=127.0.0.1 port=5001 ! " # pipe="$pipe speexdec ! queue ! alsasink sync=false " import sys,os import gobject import pygst pygst.require("0.10") import gst class server (object): def __init__(self): # start with video self.pipe = gst.Pipeline("player") # video source vidsource = gst.element_factory_make("udpsrc", "vidsource") vidsource.set_property("port", 5000) vidbin = self.buildvid() self.pipe.add(vidbin) vidsource.link(vidbin) self.pipe.set_state(gst.STATE_PLAYING) def buildvid(self): bin = gst.Bin("vidbin") queue = gst.element_factory_make("queue") smokedec = gst.element_factory_make("smokedec") vidsink = gst.element_factory_make("autovideosink") bin.add(queue, smokedec, vidsink) gst.element_link_many(queue, smokedec, vidsink) # ghostpad binsink = gst.GhostPad("binsink", queue.get_pad("sink")) bin.add_pad(binsink) return bin if __name__ == "__main__": server() loop = gobject.MainLoop() loop.run() ---------------------------------------------------------------------------- What is driving me slightly nuts is that if I put all the buildvid code into init and have a long series of object instatiations with add to pipeline and link it works, but as soon as I started messing with ghostpads and bin (which makes the code much more manageable when adding functionality for windows and macs) this messed up. Below is more code that works, and I cannot figure out why this and why not that. And the other thing is this does work in a commandline pipeline such as the one sitting at the top of the script in comments. I am using the same system for the sender, but I'll only include the local display. This sends an image happily, using the build_localvid bin, and linking it to the camera. ----------------------------------------------------------------------------- #!/bin/env python import sys, os import gobject import pygst pygst.require("0.10") import gst class client(object): def __init__(self): self.pipe = gst.Pipeline("sender") # Initial video input camera = gst.element_factory_make("v4l2src", "camera") vidtee = gst.element_factory_make("tee", "vidtee") self.pipe.add(camera, vidtee) camera.link(vidtee) # local video localvidbin = self.build_localvid() self.pipe.add(localvidbin) vidtee.link(localvidbin) self.pipe.set_state(gst.STATE_PLAYING) def build_localvid(self): """This bin takes a camera (video) stream and produces a live image locally.""" bin = gst.Bin("localvid") queue = gst.element_factory_make("queue") out = gst.element_factory_make("xvimagesink") bin.add(out, queue) queue.link_pads("src", out, "sink") binsink = gst.GhostPad("binsink", queue.get_pad("sink")) bin.add_pad(binsink) return bin if __name__ == '__main__': client() loop = gobject.MainLoop() loop.run() --------------------------------------------------------------------------- This does the correct thing, and I get a live image from the camera. I am completely stumped on this one, and have resorted to using gst.parse_bin_from_description() with the gst-launch pipeline, but this is far from ideal from a maintenance perspective. :) I am sure I am missing something pretty obvious, but so far flailing around in my ignorance has not let me stumble on a solution. Thanks for any help, Rohan From rohan at perzonae.com Wed Sep 23 13:51:07 2009 From: rohan at perzonae.com (Rohan) Date: Wed, 23 Sep 2009 13:51:07 +0200 Subject: [gst-devel] I am a signed up and confirmed member Message-ID: <4ABA0BAB.1030302@perzonae.com> Hi, As I know this will go to the moderator, I am hoping this will go through. I am a signed up and confirmed member of this list. Is there some reason that the system is not seeing this? I am receiving mails from the list in my mailbox, so I do not see why my emails are having to go through the moderator. Thanks for the impending enlightenment, Rohan From sandmann at daimi.au.dk Thu Sep 24 20:58:13 2009 From: sandmann at daimi.au.dk (Soeren Sandmann) Date: 24 Sep 2009 20:58:13 +0200 Subject: [gst-devel] [cairo] Using cairo/pixman for raw video in GStreamer In-Reply-To: <4AAE84B3.7070203@gmail.com> References: <4AAE84B3.7070203@gmail.com> Message-ID: Bill Spitzak writes: > In general I consider tiled apis to make things unnecessarily > complicated. The majority of cairo input is packed into an array. You > either need to require images to be padded out to a multiple of tile > size, or you need to greatly complicate things with "partial tiles" > with whatever code is needed to avoid ever addressing the non-existent > parts of the tiles. What I'm proposing is not to actually *store* the images in tiles, but simply to *access* them in a tiled pattern. So I'm not proposing any externally visible tiled *API* for now. (Though I think support for tiled storage may also be interesting for various reasons). > > Aside from hopefully solving the subsampling problem, tiles would also > > have better cache behavior for rotated or filtered sources. > > No the performance is TERRIBLE for filters. A filter near the edge of > a tile will require an entire neighboring tile. In scanlines the > filter always gets only the exact input scanlines needed. Consider processing an image with a 9x9 filter kernel. If you process it on a scanline by scanline basis, you will need to keep 9 scanlines in flight at the same time. This is more than will typically fit in L1, so if the cache replacement policy is Least Recently Used, then each processed cacheline will cause nine cache misses. So processing 32 scanlines of 64 cache lines causes 32 * 64 * 9 = 18432 cache misses. On the other hand processing 32 tiles of 32x32 pixels causes a total of 32 tiles times 32 + 8 rows times 2 cachelines = 2560 misses. > Tiles do help for rotation of giant images, and for drawing a section > out of the center of an image. But neither of these are common > operations for Cairo, which really wants to draw images that are > smaller than the screen fast. I don't see why tiles don't help for small rotations too. Cache lines are pretty small. Soren From aruiz at gnome.org Wed Sep 30 16:28:18 2009 From: aruiz at gnome.org (Alberto Ruiz) Date: Wed, 30 Sep 2009 15:28:18 +0100 Subject: [gst-devel] [cairo] Using cairo/pixman for raw video in GStreamer In-Reply-To: References: Message-ID: <46c5a0950909300728i7b7535b9o8790ba17a07bdb73@mail.gmail.com> 2009/9/30 Benjamin Otte : > What now? > > When talking on IRC about this I realized that there is quite a lack > of knowledge on all sides - my knowledge often isn't deep enough, > GStreamer people don't know enough about state-of-the-art video > handling and its gains and pitfalls, Cairo and X people lack knowledge > about the requirements for video playback - and this lack of knowledge > often results in preconceptions that lead to wrong decisions and make > life harder on all sides. > It would be nice if there was a way to get you people together and > actually educate each other about this process. I'd suggest a > hackfest, but I'm not sure what others think and who to approach for > funding and locations. For funding and locations you should approach someone at the board, I approached Behdad for the Gtk+ theming hackfest and I organized it in my workplace. The location would be nicer to decide once you have an idea on who's coming so that you can find a location that suits them better (and therefore it'll make it cheaper to do). As the mozilla foundation is also involved in both Cairo and video these days I'm pretty sure we can find a way for them to support the hackfest as well. So for the first step, try to come up with a list of people that would be interested in going to such hackfest. BTW, Great work!!! I can't wait to see some results. -- Cheers, Alberto Ruiz From thiagossantos at gmail.com Wed Sep 30 19:52:37 2009 From: thiagossantos at gmail.com (thiagoss) Date: Wed, 30 Sep 2009 14:52:37 -0300 Subject: [gst-devel] using the tuner interface from python In-Reply-To: <1254327454.4470.8.camel@latitude> References: <1254327454.4470.8.camel@latitude> Message-ID: On Wed, Sep 30, 2009 at 1:17 PM, Dick Marinus wrote: > Hi all, > > At April 30, 2006 Fredrik posted a message about using the tuner > interface from Python and I'm experiencing the same issue: > > $ python > >>> import gst > >>> source = gst.element_factory_make("v4l2src",name="Pinnacle PCTV > USB2.0") > >>> print source.get_channel() > > (.:13363): GStreamer-CRITICAL **: gst_implements_interface_cast: > assertion `gst_element_implements_interface (GST_ELEMENT (from), > iface_type)' failed > > ** (.:13363): CRITICAL **: gst_tuner_get_channel: assertion > `GST_IS_TUNER (tuner)' failed > > I've tried to set the pipeline queue to STATE_PLAYING but I still have > the same error. > > I guess something is broken at the Python bindings. Could someone please > help me? > I had this same problem this week on a toy project and thaytan sent me a python script that uses GstTuner interface from v4l2src, but I still haven't looked deeply at it. I'll do it tonight and I'll update here with my findings. One thing I noticed, though, is that GstV4l2Src is using a subclass of GstTunerChannel that is not mapped/exported in the python bindings (but I don't know much about bindings and thaytan said it worked for him, so I believe I'm doing something wrong in my code). > > I'm using: > gst-python=0.10.16 > gstreamer=0.10.24 > gst-plugins-good=0.10.16 > gst-plugins-base=0.10.24 > > Thanks in advance, > Dick > > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Thiago Sousa Santos -------------- next part -------------- An HTML attachment was scrubbed... URL: From kvn12345 at yahoo.com Wed Sep 30 21:40:38 2009 From: kvn12345 at yahoo.com (Khanh Nguyen) Date: Wed, 30 Sep 2009 12:40:38 -0700 (PDT) Subject: [gst-devel] qtmux Message-ID: <69496.34876.qm@web31004.mail.mud.yahoo.com> Hi, Does anybody know if qtmux accepts a buffer of one frame or can it be a buffer with multiple frames in them? thanks, --Khanh -------------- next part -------------- An HTML attachment was scrubbed... URL: From kvn12345 at yahoo.com Wed Sep 30 21:50:01 2009 From: kvn12345 at yahoo.com (Khanh Nguyen) Date: Wed, 30 Sep 2009 12:50:01 -0700 (PDT) Subject: [gst-devel] qtmux In-Reply-To: <69496.34876.qm@web31004.mail.mud.yahoo.com> References: <69496.34876.qm@web31004.mail.mud.yahoo.com> Message-ID: <368815.49586.qm@web31006.mail.mud.yahoo.com> forgot to mention that this is for audio. I'm trying to mux raw aac-lc to an mp4 container. thanks, --Khanh ________________________________ From: Khanh Nguyen To: gstreamer-devel at lists.sourceforge.net Sent: Wednesday, September 30, 2009 12:40:38 PM Subject: [gst-devel] qtmux Hi, Does anybody know if qtmux accepts a buffer of one frame or can it be a buffer with multiple frames in them? thanks, --Khanh -------------- next part -------------- An HTML attachment was scrubbed... URL: From msmith at xiph.org Wed Sep 30 22:14:57 2009 From: msmith at xiph.org (Michael Smith) Date: Wed, 30 Sep 2009 13:14:57 -0700 Subject: [gst-devel] qtmux In-Reply-To: <368815.49586.qm@web31006.mail.mud.yahoo.com> References: <69496.34876.qm@web31004.mail.mud.yahoo.com> <368815.49586.qm@web31006.mail.mud.yahoo.com> Message-ID: <3c1737210909301314r5e8097a9l7cde8b25350e7b8@mail.gmail.com> You will certainly need it in separate buffers per audio frame. You can use aacparse, probably. Mike On Wed, Sep 30, 2009 at 12:50 PM, Khanh Nguyen wrote: > forgot to mention that this is for audio.? I'm trying to mux raw aac-lc to > an mp4 container. > > thanks, > --Khanh > > ________________________________ > From: Khanh Nguyen > To: gstreamer-devel at lists.sourceforge.net > Sent: Wednesday, September 30, 2009 12:40:38 PM > Subject: [gst-devel] qtmux > > Hi, > Does anybody know if qtmux accepts a buffer of one frame or can it be a > buffer with multiple frames in them? > > thanks, > --Khanh > > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry? Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > From ylatuya at gmail.com Wed Sep 30 22:43:07 2009 From: ylatuya at gmail.com (Andoni Morales) Date: Wed, 30 Sep 2009 22:43:07 +0200 Subject: [gst-devel] creating bins and ghostpads In-Reply-To: <4ABA0AC0.7090909@perzonae.com> References: <4ABA0AC0.7090909@perzonae.com> Message-ID: <772db3280909301343y3c30474ra06f33ec5361e3b5@mail.gmail.com> 2009/9/23 Rohan : > Hi all, > > I am obviously doing something wrong here, but cannot figure out > what. > > Essentially the pipeline is the same as the receiver but I have broken > it into bins. ?I seem to be having problems with the ghostpad, because > I get this error: > > (stream_receiver.py:1914): GStreamer-WARNING **: Trying to connect elements that > don't share a common ancestor: vidsource and vidbin > > Traceback (most recent call last): > ? File "stream_receiver.py", line 118, in > ? ? server() > ? File "stream_receiver.py", line 27, in __init__ > ? ? vidsource.link(vidbin) > gst.LinkError: failed to link vidsource with vidbin > > Here is the code: > > ---------------------------------------------------------------------- > #!/bin/env python > > # The gst pipeline > # pipe="udpsrc port=5000 ! smokedec ! autovideosink" > # # audio > # pipe="$pipe tcpclientsrc host=127.0.0.1 port=5001 ! " > # pipe="$pipe speexdec ! queue ! alsasink sync=false " > > import sys,os > import gobject > import pygst > pygst.require("0.10") > import gst > > class server (object): > ? ? ? ?def __init__(self): > ? ? ? ? ? ? ? ?# start with video > > ? ? ? ? ? ? ? ?self.pipe = gst.Pipeline("player") > > ? ? ? ? ? ? ? ?# video source > ? ? ? ? ? ? ? ?vidsource = gst.element_factory_make("udpsrc", "vidsource") > ? ? ? ? ? ? ? ?vidsource.set_property("port", 5000) > > ? ? ? ? ? ? ? ?vidbin = self.buildvid() > ? ? ? ? ? ? ? ?self.pipe.add(vidbin) > ? ? ? ? ? ? ? ?vidsource.link(vidbin) You need to add vidsource to the pipeline prior to link it to any other element. That's why GStreamer is complaining about vidsource and vidbin not sharing a common ancestor. Andoni > > > ? ? ? ? ? ? ? ?self.pipe.set_state(gst.STATE_PLAYING) > > ? ? ? ?def buildvid(self): > ? ? ? ? ? ? ? ?bin = gst.Bin("vidbin") > ? ? ? ? ? ? ? ?queue = gst.element_factory_make("queue") > ? ? ? ? ? ? ? ?smokedec = gst.element_factory_make("smokedec") > ? ? ? ? ? ? ? ?vidsink = gst.element_factory_make("autovideosink") > ? ? ? ? ? ? ? ?bin.add(queue, smokedec, vidsink) > ? ? ? ? ? ? ? ?gst.element_link_many(queue, smokedec, vidsink) > > ? ? ? ? ? ? ? ?# ghostpad > ? ? ? ? ? ? ? ?binsink = gst.GhostPad("binsink", queue.get_pad("sink")) > ? ? ? ? ? ? ? ?bin.add_pad(binsink) > ? ? ? ? ? ? ? ?return bin > > if __name__ == "__main__": > ? ? ? ?server() > ? ? ? ?loop = gobject.MainLoop() > ? ? ? ?loop.run() > > ---------------------------------------------------------------------------- > > What is driving me slightly nuts is that if I put all the buildvid code > into init and have a long series of object instatiations with add to > pipeline and link it works, but as soon as I started messing with > ghostpads and bin (which makes the code much more manageable when > adding functionality for windows and macs) this messed up. ?Below is > more code that works, and I cannot figure out why this and why not > that. > > And the other thing is this does work in a commandline pipeline such > as the one sitting at the top of the script in comments. > > I am using the same system for the sender, but I'll only include the > local display. ?This sends an image happily, using the build_localvid > bin, and linking it to the camera. > > ----------------------------------------------------------------------------- > #!/bin/env python > > import sys, os > import gobject > import pygst > pygst.require("0.10") > import gst > > class client(object): > > ? ? ? ?def __init__(self): > ? ? ? ? ? ? ? ?self.pipe = gst.Pipeline("sender") > > ? ? ? ? ? ? ? ?# Initial video input > ? ? ? ? ? ? ? ?camera = gst.element_factory_make("v4l2src", "camera") > ? ? ? ? ? ? ? ?vidtee = gst.element_factory_make("tee", "vidtee") > ? ? ? ? ? ? ? ?self.pipe.add(camera, vidtee) > ? ? ? ? ? ? ? ?camera.link(vidtee) > > ? ? ? ? ? ? ? ?# local video > ? ? ? ? ? ? ? ?localvidbin = self.build_localvid() > ? ? ? ? ? ? ? ?self.pipe.add(localvidbin) > ? ? ? ? ? ? ? ?vidtee.link(localvidbin) > > ? ? ? ? ? ? ? ?self.pipe.set_state(gst.STATE_PLAYING) > > ? ? ? ?def build_localvid(self): > ? ? ? ? ? ? ? ?"""This bin takes a camera (video) stream and produces a live image locally.""" > ? ? ? ? ? ? ? ?bin = gst.Bin("localvid") > ? ? ? ? ? ? ? ?queue = gst.element_factory_make("queue") > ? ? ? ? ? ? ? ?out = gst.element_factory_make("xvimagesink") > ? ? ? ? ? ? ? ?bin.add(out, queue) > ? ? ? ? ? ? ? ?queue.link_pads("src", out, "sink") > ? ? ? ? ? ? ? ?binsink = gst.GhostPad("binsink", queue.get_pad("sink")) > ? ? ? ? ? ? ? ?bin.add_pad(binsink) > ? ? ? ? ? ? ? ?return bin > > if __name__ == '__main__': > ? ? ? ?client() > ? ? ? ?loop = gobject.MainLoop() > ? ? ? ?loop.run() > > --------------------------------------------------------------------------- > > This does the correct thing, and I get a live image from the camera. > > I am completely stumped on this one, and have resorted to using > gst.parse_bin_from_description() with the gst-launch pipeline, but > this is far from ideal from a maintenance perspective. :) > > I am sure I am missing something pretty obvious, but so far flailing > around in my ignorance has not let me stumble on a solution. > > Thanks for any help, > > Rohan > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > -- Andoni Morales Alastruey LongoMatch:The Digital Coach http://www.longomatch.ylatuya.es From startoftext at gmail.com Wed Sep 30 22:59:42 2009 From: startoftext at gmail.com (James Pearson) Date: Wed, 30 Sep 2009 15:59:42 -0500 Subject: [gst-devel] adding tee to alsasink causing problems In-Reply-To: <4AC1E7A2.9050806@perzonae.com> References: <9830AAEA-6066-4E4D-AA3E-EFFF3B9360FF@gmail.com> <4AC1E7A2.9050806@perzonae.com> Message-ID: Thanks for the help. It turns out that the problem was actually related to older sound card drivers in ubuntu. I also use gentoo and with its newer drivers there is not this problem. I switched to oss sink and problems in ubuntu went away. Some times with linux it pays to have older more ripe hardware. Thanks for you input. -James Pearson- startoftext at gmail.com 214-538-8929 On Sep 29, 2009, at 5:55 AM, Rohan wrote: > Hi James, > > I am no expert but could you put the actual gst-launch commands you > are using, because it is not clear here where you use the tee twice, > for either. > > I have been meddling with splitting audio/video but instead of for > saving to file, sending an image to the sender, as well as to another > receiving computer. > > I am guessing this might give you what you want. You do not seem to > be naming the tee, and calling it again. > > gst-launch-0.10 v4l2src ! tee name=tee ! tee. ! queue ! \ > video/x-raw-yuv,width=320,height=240 ! ffmpegcolorspace ! > ffenc_flv ! \ > ffmux_flv ! filesink location=/path/to/file tee. ! queue ! xvimagesink > > I have not tried the same with the sound, but that should get your > video recording and displaying (nota bene: not tested, just an off the > top of my head guess) > > For the sound the use of the tee should do the business. I have not > messed with mux(es) yet, so I could have gone wrong there, but the tee > stuff should probably resemble the above. You name the tee, and then > refer to it later in the commandline by name. > > Hope this helps, > > Rohan > > James Pearson wrote: >> So here is my pipeline. The point is to record audio and video and >> display it at the same time. >> >> v4l2src -> tee -> video/x-raw-yuv,width=320,height=240 -> >> ffmpegcolorspace -> ffenc_flv -> ffmux_flv -> filesink >> -> xvimagesink >> >> alsasrc -> tee > audioresample -> audio-x-raw-int,rate=44100 - -> >> audioconvert -> lame -> ffmux_flv (same mux as above) >> -> alsasink >> >> I have been using this pipeline minus the tee for live playback >> (alsasink,xvimagesink) for some time and it is working well. The >> video >> part, even with the live display works ok but when i add in the tee >> and alsasink the pipeline gets stuck with nothing going to disk and >> the video displayed is stuck. Even just adding the alsasink to the >> pipeline without connecting it causes this result. Any suggestions? I >> already tried adding queues to almost everything. >> >> Thanks >> >> -James- >> >> >> >> >> >> ------------------------------------------------------------------------------ >> Come build with us! The BlackBerry® Developer Conference in SF, >> CA >> is the only developer event you need to attend this year. Jumpstart >> your >> developing skills, take BlackBerry mobile applications to market >> and stay >> ahead of the curve. Join us from November 9-12, 2009. Register >> now! >> http://p.sf.net/sfu/devconf >> _______________________________________________ >> gstreamer-devel mailing list >> gstreamer-devel at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart > your > developing skills, take BlackBerry mobile applications to market and > stay > ahead of the curve. Join us from November 9-12, 2009. Register > now! > http://p.sf.net/sfu/devconf > _______________________________________________ > gstreamer-devel mailing list > gstreamer-devel at lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel