Search Results

Search found 61 results on 3 pages for 'gst'.

Page 1/3 | 1 2 3  | Next Page >

  • Error installing gst-python through Homebrew

    - by hrr
    I'm trying to install gst-python and it errors with: brew install gst-python Warning: Your Xcode (4.2.1) is outdated Please install Xcode 4.5.2. ==> Downloading http://gstreamer.freedesktop.org/src/gst-python/gst-python-0.10. Already downloaded: /Library/Caches/Homebrew/gst-python-0.10.22.tar.bz2 ==> ./configure --prefix=/usr/local/Cellar/gst-python/0.10.22 configure: set WARNING_CFLAGS to -Wall -Wdeclaration-after-statement -Wpointer-arith configure: set ERROR_CFLAGS to checking for valgrind... no checking for libraries required to embed python... no configure: error: could not find Python lib This is all after I removed the default OSX python and re-installed it with Homebrew, I've edited /etc/paths and python is working well. I just can't install the package above (gst, pygtk is successfully installed). i was told that I need to install python-devel but I have no idea where from. I'm using OSX Lion 10.7.5 With Python 2.7.3

    Read the article

  • 'module' object has no attribute 'element_make_factory'

    - by Ronan Dejhero
    i have this code : import pygst import st, pygtk player_name = gst.element_make_factory("playbin", "Multimedia Player") player_name.set_property("uri", "../media/alert.mp3") player_name.set_state(gst.PLAYING) it keeps throwing me the following error : player_name = gst.element_make_factory("playbin", "Multimedia Player") AttributeError: 'module' object has no attribute 'element_make_factory' nay way to solve this and why is this happening ? if i print gst i get the following : <module 'gst' from '/usr/lib/python2.7/dist-packages/gst-0.10/gst/__init__.pyc'> so it is a module !

    Read the article

  • Alpha plugin in GStreamer not working

    - by Miguel Escriva
    Hi! I'm trying to compose two videos, and I'm using the alpha plug-in to make the white color transparent. To test the alpha plug-in I'm creating the pipeline with gst-launch. The first test I done was: gst-launch videotestsrc pattern=smpte75 \ ! alpha method=custom target-r=255 target-g=255 target-b=255 angle=10 \ ! videomixer name=mixer ! ffmpegcolorspace ! autovideosink \ videotestsrc pattern=snow ! mixer. and it works great! Then I created two videos with those lines: gst-launch videotestsrc pattern=snow ! ffmpegcolorspace ! theoraenc ! oggmux ! filesink location=snow.ogv gst-launch videotestsrc pattern=smpte75 ! ffmpegcolorspace ! theoraenc ! oggmux ! filesink location=bars75.ogv And changed the videotestsrc to a filesrc and it continues working gst-launch filesrc location=bars75.ogv ! decodebin2 \ ! alpha method=custom target-r=255 target-g=255 target-b=255 angle=10 \ ! videomixer name=mixer ! ffmpegcolorspace ! autovideosink \ filesrc location=snow.ogv ! decodebin2 ! alpha ! mixer. But, when I use the ideo I want to compose, I'm not able to make the white color transparent gst-launch filesrc location=video.ogv ! decodebin2 \ ! alpha method=custom target-r=255 target-g=255 target-b=255 angle=10 \ ! videomixer name=mixer ! ffmpegcolorspace ! autovideosink \ filesrc location=snow.ogv ! decodebin2 ! alpha ! mixer. Can you help me? Any idea what is happening? I'm using GStreamer 0.10.28 You can download the test videos from here: http://polimedia.upv.es/pub/gst/gst.zip Thanks in advance, Miguel Escriva

    Read the article

  • Python + QT + Gstreamer

    - by Ptterb
    Hi everyone, I'm working with PyQt and trying to get video from a webcam to play within a QT widget. I've found tutorials for C and Qt, and for python and gtk, but NOTHING for this combo of pyQt and gstreamer. Anybody get this working? This plays the video fine, but in a separate window: self.gcam = gst.parse_launch('v4l2src device=/dev/video0 ! autovideosink') self.gcam.set_state(gst.STATE_PLAYING) what I need is to get the overlay working so it's displayed within a widget on my GUI. Thanks, Gurus of the internet! ok, so I've gotten a lot farther, but still in need of some help. I'm actually writing this for Maemo, but the following code works fine on my linux laptop: class Vid: def __init__(self, windowId): self.player = gst.Pipeline("player") self.source = gst.element_factory_make("v4l2src", "vsource") self.sink = gst.element_factory_make("autovideosink", "outsink") self.source.set_property("device", "/dev/video0") self.scaler = gst.element_factory_make("videoscale", "vscale") self.window_id = None self.windowId = windowId self.player.add(self.source, self.scaler, self.sink) gst.element_link_many(self.source,self.scaler, self.sink) bus = self.player.get_bus() bus.add_signal_watch() bus.enable_sync_message_emission() bus.connect("message", self.on_message) bus.connect("sync-message::element", self.on_sync_message) def on_message(self, bus, message): t = message.type if t == gst.MESSAGE_EOS: self.player.set_state(gst.STATE_NULL) elif t == gst.MESSAGE_ERROR: err, debug = message.parse_error() print "Error: %s" % err, debug self.player.set_state(gst.STATE_NULL) def on_sync_message(self, bus, message): if message.structure is None: return message_name = message.structure.get_name() if message_name == "prepare-xwindow-id": win_id = self.windowId assert win_id imagesink = message.src imagesink.set_property("force-aspect-ratio", True) imagesink.set_xwindow_id(win_id) def startPrev(self): self.player.set_state(gst.STATE_PLAYING) print "should be playing" vidStream = Vid(wId) vidStream.startPrev() where wId is the window id of the widget im trying to get to display the output in. When I run this on the N900, the screen goes black and blinks. Any ideas? I'm dying here!

    Read the article

  • Python and Gstreamer

    - by Seif Sallam
    hi, I'm creating a streaming application, using GStreamer with TCP pipeline, and i implemented start, pause, and stop. but the problem is, that i can't seek, i tried to change the playback value from the server side, then i tried on the client side, and Finally tried to change the value on both at the same time, but in all cases it doesn't work. and I even tried to pause the playback then continue but nothing happens. I'm having this problem with the seek and the volume. Any help please, I searched everywhere but i couldn't find anything that worked. this is the code that i use for seeking self.pipeline.seek_simple(gst.FORMAT_TIME, gst.SEEK_FLAG_FLUSH, time)

    Read the article

  • GNU Smalltalk text interface hard to use

    - by None
    I built GNU Smalltalk from source on my Mac because I couldn't get it working using fink and I found VMs like Squeak hard to understand. When I run the gst command it works fine, but unlike command line interfaces such the Python and Lua ones, it is hard to use because when I use the left or right arrow keys, I want the cursor to move left or right, but instead it inserts text like "^[[D". I understand why it does this but is there any way to fix it so it is more usable?

    Read the article

  • ruby-gstreamer doesn't send EOS message

    - by Cheba
    I've managed to make it play sound but it never gets EOS message. And thus script never exits. require 'gst' main_loop = GLib::MainLoop.new pipeline = Gst::Pipeline.new "audio-player" source = Gst::ElementFactory.make "filesrc", "file-source" source.location = "/usr/share/sounds/gnome/default/alerts/bark.ogg" decoder = Gst::ElementFactory.make "decodebin", "decoder" conv = Gst::ElementFactory.make "audioconvert", "converter" sink = Gst::ElementFactory.make "alsasink", "output" pipeline.add source, decoder, conv, sink source >> decoder conv >> sink decoder.signal_connect "pad-added" do |element, pad, data| pad >> conv['sink'] end pipeline.bus.add_watch do |bus, message| puts "Message: #{message.inspect}" case message.type when Gst::Message::Type::ERROR puts message.structure["debug"] main_loop.quit when Gst::Message::Type::EOS puts 'End of stream' main_loop.quit end end pipeline.play begin puts 'Running main loop' main_loop.run ensure puts 'Shutting down main loop' pipeline.stop end

    Read the article

  • Images or files in GNU Smalltalk?

    - by Marc
    Hi, I'm new to Smalltalk. I think I understand the basics of the language and now want to start with GNU Smalltalk (as it's free and has bindings for GTK). As I'm coming from the PHP and Java-Corner, I'm not familiar with the concept of the Smalltalk images. And I even read now, that you don't need to use images in GNU Smalltalk. Now I'm confused ;-) So is it possible to work with files and to include the classes I need with the PackageLoader class? I would be happy when I could use my favourite texteditor (vim) for coding instead of an IDE, too ;-) Please enlight me :-)

    Read the article

  • Audio decoding delay when changing the audio language

    - by mahendiran.b
    My gstreamer Pipeline is like this Approach1 --------------input-selector->Queue->AduioParser->AudioSink | Souphttpsrc->tsdemux-->| | --------------- Queue->videoParser->videoSink In this approach 1, there is a delay in audio decoding when I toggle between various audio language. Approach2 ------ input-selector-> Queue->AduioParser->AudioSink | Souphttpsrc->tsdemux---multiqueue>| | ------- Queue->videoParser->VideoSink But there is no delay is observed in approach2. Can anyone please explain the reason behind this ? what is the specialty of multiqueue here?

    Read the article

  • Alpha plugin in GStreamer not working

    - by Miguel Escriva
    Hi! I'm trying to compose two videos, and I'm using the alpha plug-in to make the white color transparent. To test the alpha plug-in I'm creating the pipeline with gst-launch. The first test I done was: gst-launch videotestsrc pattern=smpte75 \ ! alpha method=custom target-r=255 target-g=255 target-b=255 angle=10 \ ! videomixer name=mixer ! ffmpegcolorspace ! autovideosink \ videotestsrc pattern=snow ! mixer. and it works great! Then I created two videos with those lines: gst-launch videotestsrc pattern=snow ! ffmpegcolorspace ! theoraenc ! oggmux ! filesink location=snow.ogv gst-launch videotestsrc pattern=smpte75 ! ffmpegcolorspace ! theoraenc ! oggmux ! filesink location=bars75.ogv And changed the videotestsrc to a filesrc and it continues working gst-launch filesrc location=bars75.ogv ! decodebin2 \ ! alpha method=custom target-r=255 target-g=255 target-b=255 angle=10 \ ! videomixer name=mixer ! ffmpegcolorspace ! autovideosink \ filesrc location=snow.ogv ! decodebin2 ! alpha ! mixer. But, when I use the ideo I want to compose, I'm not able to make the white color transparent gst-launch filesrc location=video.ogv ! decodebin2 \ ! alpha method=custom target-r=255 target-g=255 target-b=255 angle=10 \ ! videomixer name=mixer ! ffmpegcolorspace ! autovideosink \ filesrc location=snow.ogv ! decodebin2 ! alpha ! mixer. Can you help me? Any idea what is happening? I'm using GStreamer 0.10.28 You can download the test videos from here: http://polimedia.upv.es/pub/gst/gst.zip Thanks in advance, Miguel Escriva

    Read the article

  • gstreamer libary unresolved include

    - by user300990
    Hi I wrote a simple C code with gstreamer libs( gstreamer example code manual ref ) The gstreamer headers are into /usr/include/gstreamer-0.10/gst into my C code I wrote: include "gstreamer-0.10/gst/gst.h" I have this error: there are unresolved includes inside How to solve the problem? I thank you...

    Read the article

  • Python+Windows+GStreamer = Impossible (for me)?

    - by james
    Hi :) essentially, my problem is, ater a long time searching, finding only this other question, i decided i was just going to ask my own... i am on windows 7 with python 2.6 and ossbuild GStreamer, but i am trying to get the python binding for it, and struggling. i have got gst-python from http://gstreamer.freedesktop.org/src/gst-python/ but as my eyes and research tell me, it does not work in the setup.py way, and the other question has a link to a site that he says has a binary avaiable, which no longer does, http://www.gstreamer-winbuild.ylatuya.es/doku.php?id=download and even the sdk is gone, and ossbuild dont seem to have anything useful either. so essentially, my question is, can anyone tell me a method, however convoluted, of getting my setup so if i write a script for (py)gst, with an import gst it will work? not the best of explanations... im tired k? xxx :)

    Read the article

  • GNU Smalltalk package

    - by Peter
    I've installed the GNU Smalltalk package and can get to the SmallTalk command line with the command 'gst'. However, I can't start the visual gst browser using the command: $ gst-browser When I try, this is what I get: peter@peredur:~$ gst-browser Object: CFunctionDescriptor new: 1 "<0x40488720>" error: Invalid C call-out gdk_colormap_get_type SystemExceptions.CInterfaceError(Smalltalk.Exception)>>signal (ExcHandling.st:254) SystemExceptions.CInterfaceError class(Smalltalk.Exception class)>>signal: (ExcHandling.st:161) Smalltalk.CFunctionDescriptor(Smalltalk.CCallable)>>callInto: (CCallable.st:165) GdkColormap class>>getType (GTK.star#VFS.ZipFile/Funcs.st:1) optimized [] in GLib class>>registerAllTypes (GTK.star#VFS.ZipFile/GtkDecl.st:78) Smalltalk.OrderedCollection>>do: (OrderColl.st:68) GLib class>>registerAllTypes (GTK.star#VFS.ZipFile/GtkDecl.st:78) Smalltalk.UndefinedObject>>executeStatements (GTK.star#VFS.ZipFile/GtkImpl.st:1078) Object: CFunctionDescriptor new: 1 "<0x404a7c28>" error: Invalid C call-out gtk_window_new SystemExceptions.CInterfaceError(Exception)>>signal (ExcHandling.st:254) SystemExceptions.CInterfaceError class(Exception class)>>signal: (ExcHandling.st:161) CFunctionDescriptor(CCallable)>>callInto: (CCallable.st:165) GTK.GtkWindow class>>new: (GTK.star#VFS.ZipFile/Funcs.st:1) VisualGST.GtkDebugger(VisualGST.GtkMainWindow)>>initialize (VisualGST.star#VFS.ZipFile/GtkMainWindow.st:131) VisualGST.GtkDebugger class(VisualGST.GtkMainWindow class)>>openSized: (VisualGST.star#VFS.ZipFile/GtkMainWindow.st:19) [] in VisualGST.GtkDebugger class>>open: (VisualGST.star#VFS.ZipFile/Debugger/GtkDebugger.st:16) [] in BlockClosure>>forkDebugger (DebugTools.star#VFS.ZipFile/DebugTools.st:380) [] in Process>>onBlock:at:suspend: (Process.st:392) BlockClosure>>on:do: (BlkClosure.st:193) [] in Process>>onBlock:at:suspend: (Process.st:393) BlockClosure>>ensure: (BlkClosure.st:269) [] in Process>>onBlock:at:suspend: (Process.st:370) [] in BlockClosure>>asContext: (BlkClosure.st:179) BlockContext class>>fromClosure:parent: (BlkContext.st:68) Everything hangs at this point until I hit ^C, after which, I get: Object: CFunctionDescriptor new: 1 "<0x404a7c28>" error: Invalid C call-out gtk_window_new SystemExceptions.CInterfaceError(Exception)>>signal (ExcHandling.st:254) SystemExceptions.CInterfaceError class(Exception class)>>signal: (ExcHandling.st:161) CFunctionDescriptor(CCallable)>>callInto: (CCallable.st:165) GTK.GtkWindow class>>new: (GTK.star#VFS.ZipFile/Funcs.st:1) VisualGST.GtkDebugger(VisualGST.GtkMainWindow)>>initialize (VisualGST.star#VFS.ZipFile/GtkMainWindow.st:131) VisualGST.GtkDebugger class(VisualGST.GtkMainWindow class)>>openSized: (VisualGST.star#VFS.ZipFile/GtkMainWindow.st:19) [] in VisualGST.GtkDebugger class>>open: (VisualGST.star#VFS.ZipFile/Debugger/GtkDebugger.st:16) [] in BlockClosure>>forkDebugger (DebugTools.star#VFS.ZipFile/DebugTools.st:380) [] in Process>>onBlock:at:suspend: (Process.st:392) BlockClosure>>on:do: (BlkClosure.st:193) [] in Process>>onBlock:at:suspend: (Process.st:393) BlockClosure>>ensure: (BlkClosure.st:269) [] in Process>>onBlock:at:suspend: (Process.st:370) [] in BlockClosure>>asContext: (BlkClosure.st:179) BlockContext class>>fromClosure:parent: (BlkContext.st:68) peter@peredur:~$ Is there a problem with this package?

    Read the article

  • Rhythmbox won't import or play flac files

    - by Dan Drake
    I have a new installation of 12.04 and I just copied over all my music to the ~/Music folder. Rhythmbox found all the mp3 and ogg files, but it refuses to import flac files. They simply do not appear in my music library. If I start Rhythmbox on the command line and try to import a folder that contains flac files, absolutely nothing happens. Nothing is imported; no error messages. I have all the dependencies for Rhythmbox installed, along with all the suggested and recommended packages. I can play a flac file with gst-launch-0.10 and gst-typefind-0.10 correctly identifies flac files as audo/x-flac. Why does Rhythmbox refuse to see flac files? What can I do to find out what is happening?

    Read the article

  • Why can't I access hw:1,0 until gstreamer-properties is run once?

    - by Shadd
    Not necessarily an Ubuntu-specific question but I wasn't sure where else to ask. I have an AverMedia DVD EZMaker 7 which plugs into USB and works well on Ubuntu 12.04. I downloaded and installed the official drivers without error. However, when I try: gst-launch alsasrc device="hw:1,0" ! alsasink device="hw:0,0" it tells me: Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstAudioSrcClock which is the normal output, however there is no audible sound. Trying the command again doesn't help. If I run gstreamer-properties and close it right away (don't need to touch any controls), THEN the gst-launch command works. If I unplug the device and plug it back in or restart the computer, I have to run gstreamer-properties again. What is gstreamer-properties doing that enables the audio?

    Read the article

  • Needed environment for building gstreamer plugins in Windows

    - by utnapistim
    Hi, I've been strugling for two weeks to create an environment for building a gstreamer plugin on windows (needed for a songbird addon). I've installed MSYS, MinGW and Cygwin, then installed GStreamer OSSBuild, and I also downloaded the sources for Songbird, which come with their own precompiled version of gstreamer. I was unable to run gst-inspect (or any other gstreamer applications) from the songbird sources and I figured I will settle for OSSBuild. When following the instructions for building a GST plugin (found here) through, cygwin will not recognize the OSSBuild and the build fails when running autogen, with the following error: checking for GST... no configure: error: You need to install or upgrade the GStreamer development packages on your system. On debian-based systems these are libgstreamer0.10-dev and libgstreamer-plugins-base0.10-dev. on RPM-based systems gstreamer0.10-devel, libgstreamer0.10-devel or similar. The minimum version required is 0.10.16. configure failed I could also not use MSYS or MinGW as they are unable to run autogen at all. I understand that cygwin should have it's own gstreamer development packages but I couldn't find how to install them. My question: How do I install the gstreamer packages in cygwin or how do I build using cygwin with the OSSBuild dependencies? In short, how do I get an environment where I can build a gstreamer plugin under windows?

    Read the article

  • How do I set up live audio streams to a DLNA compliant device?

    - by Takkat
    Is there a way to stream the live output of the soundcard from our 12.04.1 LTS amd64 desktop to a DLNA-compliant external device in our network? Selecting media content in shared directories using Rygel, miniDLNA, and uShare is always fine - but so far we completely failed to get a live audio stream to a client via DLNA. Pulseaudio claims to have a DLNA/UPnP media server that together with Rygel is supposed to do just this. But we were unable to get it running. We followed the steps outlined in live.gnome.org, this answer here, and also in another similar guide. As soon as we select the local audio device, or our GST-Launch stream in the DLNA client Rygel displays the following message and the client states it reached the end of the playlist: (rygel:7380): Rygel-WARNING **: rygel-http-request.vala:97: Invalid seek request This is how we configured GST-Launch in rygel.conf: [GstLaunch] enabled=true launch-items=mypulseaudiosink mypulseaudiosink-title=Audio on @HOSTNAME@ mypulseaudiosink-mime=audio/x-wav mypulseaudiosink-launch=pulsesrc device=<device> ! wavpackenc For <device> we tried with the default sink name, this name appended with .monitor, and in addition with upnp-sink and upnp.monitor that was created when we selected DLNA media server from paprefs. We also tried to encode using lamemp3enc with no luck. These are our pulseaudio modules: http://paste.ubuntu.com/1202913/ These are our sinks: http://paste.ubuntu.com/1202916/ Did we miss any other additional configuration needed to get this running? Are there any other alternatives for sending the audio of our soundcard as live stream to a DLNA client?

    Read the article

  • There's A Virtual Developer Day in Your Future

    - by OTN ArchBeat
    What are Virtual Developer Days? You really should know this by now. OTN Virtual Developer Days are online events created specifically for developers and architects, with a focus on no-fluff technical presentations, hands-on labs, and expert Q&A to sharpen your technical skills and bring you up to speed on the latest information on Oracle products and practical best practices for their use. The best part about OTN Virtual Developer Days is that you don't have to pack a suitcase or stand in line at an airport waiting for someone pat you down. Instead, you stay where you are, flip open your laptop, and prepare your brain for a massive skills injection. In the next few weeks you'll have two such chances to ramp up your skills. On Tuesday November 5, 2013 Harnessing the Power of Oracle WebLogic and Oracle Coherence will guide you through tooling updates and best practices for developing applications with WebLogic and Coherence as target platforms. This two-track event covers app design and development (Track 1) and building, deploying, and managing applications (Track 2). Each track includes three presentations plus a hands-on lab. [9am-1pm PT / 12pm-4pm ET / 1pm-5pm BRT] Register now This event will also be available in EMEA on December 3, 2013 {9am-1pm GMT / 1pm-5pm GST / 2:30pm-6:30 PM IST] On Tuesday November 19, 2103 Oracle ADF Development: Web, Mobile, and Beyond offers four tracks covering everything from the basics to advance skills for for application development using Oracle ADF and Oracle ADF Mobile. There are three sessions in each track, followed by hands-on labs in which try out what you've learned. [9am-1pm PT / 12pm-4pm ET/ 1pm-5pm BRT] Register now This event will also be available in APAC on Thursday November 21, 2013 [10am-1:30pm IST (India) / 12:30pm-4pm SGT (Singapore) / 3:30pm-7pm AESDT] and in EMEA on Tuesday November 26, 2013 [9am-1pm GMT / 1pm-5pm GST/ 2:30pm-6:30pm IST] Registration for both events is absolutely free. So what are you waiting for?

    Read the article

  • Registration Now Open: Virtual Developer Day, North America, APAC & Europe

    - by Tanu Sood
    Is your organization looking at developing Web or Mobile application based upon the Oracle platform?  Oracle is offering a virtual event for Developer Leads, Managers and Architects to learn more about developing Web, Mobile and beyond based on Oracle applications. This event will provide sessions that range from introductory overviews to technical deep dives covering Oracle's strategic framework for developing multi-channel enterprise applications for the Oracle platforms. Multiple tracks cover every interest and every level and include live online Q&A chats with Oracle's technical staff. For registration and information on Vortual Developer Day: Oracle ADF Development, please follow the link HERE Sign up for one of the following events below: Americas - Tuesday - November 19th / 9am to 1pm PDT / 12pm to 4pm EDT / 1pm to 5pm BRT APAC - Thursday - November 21st / 10am - 1:30pm IST (India) / 12:30pm - 4pm SGT (Singapore) / 3:30pm -7pm AESDT EMEA - Tuesday - November 26th / 9am - 1pm GMT / 1pm - 5pm GST / 2:30pm -6:30pm IST And for those interested in Cloud Application Foundation, including Weblogic and Coherence, don't forget to sign up for the following events: Americas - Tuesday, November 5, 2013 - 9 am - 1 pm PDT/ 12 pm - 4 pm EDT/ 1 pm - 5 pm BRT EMEA - December 3, 2013 - 9 a, - 1 pm GMT/ 1pm - 5pm GST/ 2:30 pm - 6:30 pm ISTThe event will guide you through tooling updates and best practices around developing applications with WebLogic and Coherence as target platforms.

    Read the article

  • magento - multiple tax rates

    - by Fiona
    I've built a magento site for a Canadian company where each state has a Retail sales tax and a federal tax rate and these rates differ! So, I set up the rates, and the Tax is being calculated correctly, ie the sum of the two tax rates is being calculated correctly. however on selecting the breakdown of tax mode in the admin panel, it would appear that there is something wrong with my setup. Eg. Subtotal $129.99 GST/HST Quebec (5%) $ 16.25 Provincial Sales Tax Quebec (7.5%) Tax $ 16.25 Grand Total $158.23 16.25 is 12% of $129.99 so the tax figure is correct. However it should be displaying as follows: Subtotal $129.99 GST/HST Quebec (5%) $ 6.50 Provincial Sales Tax Quebec (7.5%) $ 9.75 Tax $ 16.25 Grand Total $158.23 Anyone come across this before? Have any suggestions on how to fix it? Many thanks, Fiona

    Read the article

  • Need a formula for calculating the tax portion of a total amount.

    - by pawz
    In Australia we have to advertise products with tax already added, so rather than say a product is $10 + $1 GST = $11, we normally work backwards and say "ok, total is $10, how much of that is GST ?" For example, for a $10 total, you do 10 * (1 /11) = 0.91, which is the tax component of the $10 total. My problem is I need calculate a formula for working out the taxable component when the tax rate is a variable. So far I've made this calculation although I'm not sure how correct an assertion it is: 10 * (1 / x) = 0.09 * (1 / y) where y = 10, x = 11 Basically i want to work out x on the left hand side when I know that the tax rate is 0.05 for example, which will give me a formula that I can use to calculate the taxable component of an total figure. I want a function into which I can plug in the total price and the tax rate, and get back the taxable component of the total price. I'd really appreciate the help with this as it really makes my head hurt ! :")

    Read the article

  • CUDA, more threads for same work = Longer run time despite better occupancy, Why?

    - by zenna
    I encountered a strange problem where increasing my occupancy by increasing the number of threads reduced performance. I created the following program to illustrate the problem: #include <stdio.h> #include <stdlib.h> #include <cuda_runtime.h> __global__ void less_threads(float * d_out) { int num_inliers; for (int j=0;j<800;++j) { //Do 12 computations num_inliers += threadIdx.x*1; num_inliers += threadIdx.x*2; num_inliers += threadIdx.x*3; num_inliers += threadIdx.x*4; num_inliers += threadIdx.x*5; num_inliers += threadIdx.x*6; num_inliers += threadIdx.x*7; num_inliers += threadIdx.x*8; num_inliers += threadIdx.x*9; num_inliers += threadIdx.x*10; num_inliers += threadIdx.x*11; num_inliers += threadIdx.x*12; } if (threadIdx.x == -1) d_out[blockIdx.x*blockDim.x+threadIdx.x] = num_inliers; } __global__ void more_threads(float *d_out) { int num_inliers; for (int j=0;j<800;++j) { // Do 4 computations num_inliers += threadIdx.x*1; num_inliers += threadIdx.x*2; num_inliers += threadIdx.x*3; num_inliers += threadIdx.x*4; } if (threadIdx.x == -1) d_out[blockIdx.x*blockDim.x+threadIdx.x] = num_inliers; } int main(int argc, char* argv[]) { float *d_out = NULL; cudaMalloc((void**)&d_out,sizeof(float)*25000); more_threads<<<780,128>>>(d_out); less_threads<<<780,32>>>(d_out); return 0; } Note both kernels should do the same amount of work in total, the (if threadIdx.x == -1 is a trick to stop the compiler optimising everything out and leaving an empty kernel). The work should be the same as more_threads is using 4 times as many threads but with each thread doing 4 times less work. Significant results form the profiler results are as followsL: more_threads: GPU runtime = 1474 us,reg per thread = 6,occupancy=1,branch=83746,divergent_branch = 26,instructions = 584065,gst request=1084552 less_threads: GPU runtime = 921 us,reg per thread = 14,occupancy=0.25,branch=20956,divergent_branch = 26,instructions = 312663,gst request=677381 As I said previously, the run time of the kernel using more threads is longer, this could be due to the increased number of instructions. Why are there more instructions? Why is there any branching, let alone divergent branching, considering there is no conditional code? Why are there any gst requests when there is no global memory access? What is going on here! Thanks

    Read the article

  • Java and gstreamer-java initialisation error

    - by Mark
    I am building a small app which will play streaming audio from the internet in java (mainly internet radio stations). I have decided to use the gstreamer-java library for the sound, which uses JNA. I would like to include a check in the code, to see whether the gstreamer library has been initialised. When I have left the "Gst.init()" code out (to mimic when the library has not been initialised correctly), the application throws out the following messages: (process:21888): GLib-GObject-CRITICAL **: /build/buildd/glib2.0-2.22.3/gobject/gtype.c:2458: initialization assertion failed, use IA__g_type_init() prior to this function (process:21888): GLib-CRITICAL **: g_once_init_leave: assertion `initialization_value != 0' failed The app calls the gstreamer-java library. The error messages appear but the thread continues to run, hogging the CPU. Is there any way to catch the error or to add a check to prevent it from happening? An alternative would be to put the "Gst.init()" in the main class, but I am not sure if this would always guarantee the gstreamer library is initialised.

    Read the article

  • Why does use of H264 in sender/receiver pipelines introduce just HUGE delay?

    - by Serguey Zefirov
    When I try to create pipeline that uses H264 to transmit video, I get some enormous delay, up to 10 seconds to transmit video from my machine to... my machine! This is unacceptable for my goals and I'd like to consult StackOverflow over what I (or someone else) do wrong. I took pipelines from gstrtpbin documentation page and slightly modified them to use Speex: This is sender pipeline: #!/bin/sh gst-launch -v gstrtpbin name=rtpbin \ v4l2src ! ffmpegcolorspace ! ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 \ rtpbin.send_rtp_src_0 ! udpsink host=127.0.0.1 port=5000 \ rtpbin.send_rtcp_src_0 ! udpsink host=127.0.0.1 port=5001 sync=false async=false \ udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \ pulsesrc ! audioconvert ! audioresample ! audio/x-raw-int,rate=16000 ! \ speexenc bitrate=16000 ! rtpspeexpay ! rtpbin.send_rtp_sink_1 \ rtpbin.send_rtp_src_1 ! udpsink host=127.0.0.1 port=5002 \ rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5003 sync=false async=false \ udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1 Receiver pipeline: !/bin/sh gst-launch -v\ gstrtpbin name=rtpbin \ udpsrc caps="application/x-rtp,media=(string)video, clock-rate=(int)90000, encoding-name=(string)H263-1998" \ port=5000 ! rtpbin.recv_rtp_sink_0 \ rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink \ udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false \ udpsrc caps="application/x-rtp,media=(string)audio, clock-rate=(int)16000, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110" \ port=5002 ! rtpbin.recv_rtp_sink_1 \ rtpbin. ! rtpspeexdepay ! speexdec ! audioresample ! audioconvert ! alsasink \ udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5007 sync=false async=false Those pipelines, a combination of H263 and Speex, work fine enough. I snap my fingers near camera and micropohne and then I see movement and hear sound at the same time. Then I changed pipelines to use H264 along the video path. The sender becomes: #!/bin/sh gst-launch -v gstrtpbin name=rtpbin \ v4l2src ! ffmpegcolorspace ! x264enc bitrate=300 ! rtph264pay ! rtpbin.send_rtp_sink_0 \ rtpbin.send_rtp_src_0 ! udpsink host=127.0.0.1 port=5000 \ rtpbin.send_rtcp_src_0 ! udpsink host=127.0.0.1 port=5001 sync=false async=false \ udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \ pulsesrc ! audioconvert ! audioresample ! audio/x-raw-int,rate=16000 ! \ speexenc bitrate=16000 ! rtpspeexpay ! rtpbin.send_rtp_sink_1 \ rtpbin.send_rtp_src_1 ! udpsink host=127.0.0.1 port=5002 \ rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5003 sync=false async=false \ udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1 And receiver becomes: #!/bin/sh gst-launch -v\ gstrtpbin name=rtpbin \ udpsrc caps="application/x-rtp,media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264" \ port=5000 ! rtpbin.recv_rtp_sink_0 \ rtpbin. ! rtph264depay ! ffdec_h264 ! xvimagesink \ udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false \ udpsrc caps="application/x-rtp,media=(string)audio, clock-rate=(int)16000, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110" \ port=5002 ! rtpbin.recv_rtp_sink_1 \ rtpbin. ! rtpspeexdepay ! speexdec ! audioresample ! audioconvert ! alsasink \ udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5007 sync=false async=false This is what happen under Ubuntu 10.04. I didn't noticed such huge delays on Ubuntu 9.04 - the delays there was in range 2-3 seconds, AFAIR.

    Read the article

1 2 3  | Next Page >