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  • How to fix Windows XP from muting sounds on every startup?

    - by Rookie
    This started to happen few days ago: Every time I start up my computer, the global sound volume ("Playback") is set to muted (from the checkbox), in the sound control panel. So I have to open the control panel every time i start up my computer, and click that checkbox. I have got this bug before, long time ago. I have no idea why is this happening. Any ideas how to fix this or what is causing it? I have SP 3 and most updates installed a half year ago or less.

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  • Installing a fake microphone on Windows Server.

    - by Adrian
    My application, that is running on Windows Server (which is an instance on Amazon EC2) requires Skype to be able to make phone calls. The server, of course, does not have a microphone installed and I don't need it to have one, because my application changes the input source to a wav file when the call is established. However, Skype has a strict rule that a microphone must be installed for a call to be made. Thus I want to install a fake microphone that will trick Skype's configuration. So far, I was able to start and run the Windows Sound service, which enabled all of the sound settings. Any ideas are very welcome!

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  • VirtualBox sound problem under Ubuntu

    - by VoY
    As I recently upgraded to karmic I started to see the following stuff in the logs when I run VirtualBox: Oct 30 18:14:34 apocalypse pulseaudio[2813]: alsa-source.c: Resume failed, couldn't restore original fragment settings. (Old: 65536/65536, New 1073676288/65532)io[2813]: alsa-source.c: Resume failed, couldn't restore original fragment settings. (Old: 65536/65536, New 1073676288/65532)io[2813]: alsa-source.c: Resume failed, couldn't restore original fragment settings. (Old: 65536/65536, New 1073676288/65532)io[2813]: alsa-source.c: Resume failed, couldn't restore original fragment settings. (Old: 65536/65536, New 1073676288/65532)io[2813]: alsa-source.c: Resume failed, couldn't restore original fragment settings. (Old: 65536/65536, New 1073676288/65532)io[2813]: alsa-source.c: Resume failed, couldn't restore original fragment settings. (Old: 65536/65536, New 1073676288/65532)io[2813]: alsa-source.c: Resume failed, couldn't restore original fragment settings. (Old: 65536/65536, New 1073676288/65532)io[2813]: alsa-source.c: Resume failed, couldn't restore original fragment settings. (Old: 65536/65536, New 1073676288/65532)io[2813]: alsa-source.c: Resume failed, couldn't restore original fragment settings. (Old: 65536/65536, New 1073676288/65532)io[2813]: alsa-source.c: Resume failed, couldn't restore original fragment settings. (Old: 65536/65536, New 1073676288/65532)io[2813]: alsa-source.c: Resume failed, couldn't restore original fragment settings. (Old: 65536/65536, New 1073676288/65532)io[2813]: alsa-source.c: Resume failed, couldn't restore original fragment settings. (Old: 65536/65536, New 1073676288/65532) Oct 30 18:14:34 apocalypse pulseaudio[2813]: alsa-source.c: Resume failed, couldn't restore original fragment settings. (Old: 65536/65536, New 1073676288/65532) After a while the logs grow to large sizes and fill up all of my /var partition. In VirtualBox there is an option to choose between pulseaudio and alsa for sound, but it seems to have no effect. I am using virtualbox-3.0 packages, not the ose version. My system is up to date.

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  • Sound out of sync after merging multiple mp4 files with Avidemux

    - by Goto10
    I am trying to join (merge) two or more .mp4 files together, without re-encoding. Here is what I did: Started Avidemux 2.5.5. With File-Open, selected Input1.mp4. I received this message - "H.264 detected. If the file is using B-frames as reference it can lead to a crash or stuttering. Avidemux can use another mode which is safe but YOU WILL LOOSE SOME FRAME ACCURACY. Do you want to use that mode?". I chose "No". With File-Append, selected Input2.mp4. I received the same "H.264 detected" message again and chose "No". Selected the Format to MP4 (from AVI). Saved the output file (called Output.mp4) with File-Save-Save Video. Unfortunately, when I play the Output.mp4 video in VLC, the sound is out of sync with the second video. How can I correct this?

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  • Laptop freezing every few seconds, including screen + sound

    - by zenstealth
    Just a few days ago, my Windows 7 HP dv4170us (1.76Ghz CPU, 1GB Ram) laptop started to freeze every other second where everything on screen and and sound (such as a song playing in iTunes) would just freeze until I bash it violently (without actually breaking the laptop) or wait for a couple of more seconds. I think it started one night when I noticed that a USB mouse of mine stopped working, and it displayed random "Device was not recognized" errors. I just unplugged the mouse and ignored it. Skip forward to the next day, is started freezing, and as of today I can't get my computer to not keep freezing. I tried to backup my files onto an external hdd, but it almost corrupted the drive. I ran 4 complete virus scans using MSSE and MalwareBytes (both quick and full scans), and they all came up clean. In the Task manager, the CPU usage is on a constant max, and so is the RAM (if I have just a few apps running, I only have like 30Mb of free RAM left). Also, on the outside of my laptop, right above where the CPU is located, it's very, very hot. I suspect that something is wrong internally within inside of the computer, but I'm not sure. It also does the same thing when booted into Ubuntu.Does anyone know what could be wrong with it?

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  • Sound doesn't work anymore after replacing RAM

    - by thejh
    Hello, today, I replaced one old RAM module with two newer, bigger ones, but now, the sound doesn't seem to work anymore. Already ran alsaconf and it didn't help. Output of lspci for the audio device: 00:07.0 Audio device: nVidia Corporation MCP67 High Definition Audio (rev a1) Subsystem: Giga-byte Technology Device a002 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 (500ns min, 1250ns max) Interrupt: pin A routed to IRQ 21 Region 0: Memory at f5100000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [50] Message Signalled Interrupts: Mask+ 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Masking: 00000000 Pending: 00000000 Capabilities: [6c] HyperTransport: MSI Mapping Enable+ Fixed+ Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel The audio device is onboard and has six configurable outputs, two or so are also capable of being an input (if I remember it correctly), but I don't know how to control it under linux. Does somebody know how/whether replacing the RAM could be related to my problem and/or how to fix it?

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  • Google Chrome on Linux not using ALSA for sound

    - by DarkMoon
    On my laptop, I've got an .asoundrc file that outputs sound to my USB headset. This works fine for SMplayer and Firefox. However, Google Chrome (at least, Flash-based and HTML5-based videos and HTML5-based audio in Chrome) plays through the laptop speakers instead. I've tried running Chrome from a command-line, hoping there would be some helpful output, but no such luck. I've tried looking through Google for whether Chrome even uses ALSA, or if it uses something else, but I have been unsuccessful in this. This question seems to be the same issue, but no suggestion was made. Anyone have any ideas? I'm running Gentoo with a 3.10.17 kernel, 1.0.27 ALSA utils, 2.6.5 FVWM, and 36.0.1985.143 Chrome. If you need more info, please let me know. EDIT: I've configured the USB headset as the default ALSA device. Volume levels for both headset and onboard are set and un-muted using alsamixer. My .asoundrc file is as follows. ctl.!default { type hw card Headset } pcm.dmixer { type dmix ipc_key 1024 slave { pcm { type hw card Headset } period_size 1024 buffer_size 4096 } bindings { 0 0 1 1 } } pcm.!default { type plug slave.pcm dmixer }

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  • Using jQuery and OData to Insert a Database Record

    - by Stephen Walther
    In my previous blog entry, I explored two ways of inserting a database record using jQuery. We added a new Movie to the Movie database table by using a generic handler and by using a WCF service. In this blog entry, I want to take a brief look at how you can insert a database record using OData. Introduction to OData The Open Data Protocol (OData) was developed by Microsoft to be an open standard for communicating data across the Internet. Because the protocol is compatible with standards such as REST and JSON, the protocol is particularly well suited for Ajax. OData has undergone several name changes. It was previously referred to as Astoria and ADO.NET Data Services. OData is used by Sharepoint Server 2010, Azure Storage Services, Excel 2010, SQL Server 2008, and project code name “Dallas.” Because OData is being adopted as the public interface of so many important Microsoft technologies, it is a good protocol to learn. You can learn more about OData by visiting the following websites: http://www.odata.org http://msdn.microsoft.com/en-us/data/bb931106.aspx When using the .NET framework, you can easily expose database data through the OData protocol by creating a WCF Data Service. In this blog entry, I will create a WCF Data Service that exposes the Movie database table. Create the Database and Data Model The MoviesDB database is a simple database that contains the following Movies table: You need to create a data model to represent the MoviesDB database. In this blog entry, I use the ADO.NET Entity Framework to create my data model. However, WCF Data Services and OData are not tied to any particular OR/M framework such as the ADO.NET Entity Framework. For details on creating the Entity Framework data model for the MoviesDB database, see the previous blog entry. Create a WCF Data Service You create a new WCF Service by selecting the menu option Project, Add New Item and selecting the WCF Data Service item template (see Figure 1). Name the new WCF Data Service MovieService.svc. Figure 1 – Adding a WCF Data Service Listing 1 contains the default code that you get when you create a new WCF Data Service. There are two things that you need to modify. Listing 1 – New WCF Data Service File using System; using System.Collections.Generic; using System.Data.Services; using System.Data.Services.Common; using System.Linq; using System.ServiceModel.Web; using System.Web; namespace WebApplication1 { public class MovieService : DataService< /* TODO: put your data source class name here */ > { // This method is called only once to initialize service-wide policies. public static void InitializeService(DataServiceConfiguration config) { // TODO: set rules to indicate which entity sets and service operations are visible, updatable, etc. // Examples: // config.SetEntitySetAccessRule("MyEntityset", EntitySetRights.AllRead); // config.SetServiceOperationAccessRule("MyServiceOperation", ServiceOperationRights.All); config.DataServiceBehavior.MaxProtocolVersion = DataServiceProtocolVersion.V2; } } } First, you need to replace the comment /* TODO: put your data source class name here */ with a class that represents the data that you want to expose from the service. In our case, we need to replace the comment with a reference to the MoviesDBEntities class generated by the Entity Framework. Next, you need to configure the security for the WCF Data Service. By default, you cannot query or modify the movie data. We need to update the Entity Set Access Rule to enable us to insert a new database record. The updated MovieService.svc is contained in Listing 2: Listing 2 – MovieService.svc using System.Data.Services; using System.Data.Services.Common; namespace WebApplication1 { public class MovieService : DataService<MoviesDBEntities> { public static void InitializeService(DataServiceConfiguration config) { config.SetEntitySetAccessRule("Movies", EntitySetRights.AllWrite); config.DataServiceBehavior.MaxProtocolVersion = DataServiceProtocolVersion.V2; } } } That’s all we have to do. We can now insert a new Movie into the Movies database table by posting a new Movie to the following URL: /MovieService.svc/Movies The request must be a POST request. The Movie must be represented as JSON. Using jQuery with OData The HTML page in Listing 3 illustrates how you can use jQuery to insert a new Movie into the Movies database table using the OData protocol. Listing 3 – Default.htm <!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"> <html xmlns="http://www.w3.org/1999/xhtml"> <head> <title>jQuery OData Insert</title> <script src="http://ajax.microsoft.com/ajax/jquery/jquery-1.4.2.js" type="text/javascript"></script> <script src="Scripts/json2.js" type="text/javascript"></script> </head> <body> <form> <label>Title:</label> <input id="title" /> <br /> <label>Director:</label> <input id="director" /> </form> <button id="btnAdd">Add Movie</button> <script type="text/javascript"> $("#btnAdd").click(function () { // Convert the form into an object var data = { Title: $("#title").val(), Director: $("#director").val() }; // JSONify the data var data = JSON.stringify(data); // Post it $.ajax({ type: "POST", contentType: "application/json; charset=utf-8", url: "MovieService.svc/Movies", data: data, dataType: "json", success: insertCallback }); }); function insertCallback(result) { // unwrap result var newMovie = result["d"]; // Show primary key alert("Movie added with primary key " + newMovie.Id); } </script> </body> </html> jQuery does not include a JSON serializer. Therefore, we need to include the JSON2 library to serialize the new Movie that we wish to create. The Movie is serialized by calling the JSON.stringify() method: var data = JSON.stringify(data); You can download the JSON2 library from the following website: http://www.json.org/js.html The jQuery ajax() method is called to insert the new Movie. Notice that both the contentType and dataType are set to use JSON. The jQuery ajax() method is used to perform a POST operation against the URL MovieService.svc/Movies. Because the POST payload contains a JSON representation of a new Movie, a new Movie is added to the database table of Movies. When the POST completes successfully, the insertCallback() method is called. The new Movie is passed to this method. The method simply displays the primary key of the new Movie: Summary The OData protocol (and its enabling technology named WCF Data Services) works very nicely with Ajax. By creating a WCF Data Service, you can quickly expose your database data to an Ajax application by taking advantage of open standards such as REST, JSON, and OData. In the next blog entry, I want to take a closer look at how the OData protocol supports different methods of querying data.

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  • How can I make sound work without starting X?

    - by Magnus Hoff
    I have a headless machine connected to my sound system, and I am using it to run a music playing daemon that I control over the network. (Among other things) However, I can't seem to be able to have sound come out of my speakers without running X. I am running pulse audio in a system wide instance and my daemon is not running within X. Nevertheless, when my daemon is playing music without me hearing it, I can fix it by running startx in an unrelated session. After X starts, I can hear the sound. The sound disappears again if I kill the X server. Interestingly/annoyingly, the sound also stops after X has been running for a few minutes. This could possibly be because of a screen saver of some sort, but I haven't been able to verify or falsify this theory. So my current workaround is to ssh into the box whenever I want music and startx, and restart it every fifteen minutes or so. I'd like to do better. I have been able to verify the following: Adjustments in alsamixer have no effect on this problem. The relevant output channel is never muted In alsamixer, I can see no difference between when the sound is working and when it isn't Nothing is muted in pactl list There is no difference in the output from pactl list between before starting X and after it's started. (Except the identifier of the pactl instance connected to pulse, which is different each time you run pactl) The user running the music daemon is a member of the groups audio, pulse and pulse-access The music daemon program does not report any error messages and acts as if it is playing the music like it should Some form of dbus daemon is running. ps aux|grep dbus reports dbus-daemon --system --fork --activation=upstart before and after I have started X Some details about my hardware: Motherboard: http://www.asus.com/Motherboards/AT5IONTI_DELUXE/ Sound chip: Nvidia GPU 0b HDMI/DP (from alsamixer) Using HDMI for output (Machine also has an Intel Realtek ALC887 that I am not using) Output of lsmod: Module Size Used by deflate 12617 0 zlib_deflate 27139 1 deflate ctr 13201 0 twofish_generic 16635 0 twofish_x86_64_3way 25287 0 twofish_x86_64 12907 1 twofish_x86_64_3way twofish_common 20919 3 twofish_generic,twofish_x86_64_3way,twofish_x86_64 camellia 29348 0 serpent 29125 0 blowfish_generic 12530 0 blowfish_x86_64 21466 0 blowfish_common 16739 2 blowfish_generic,blowfish_x86_64 cast5 25112 0 des_generic 21415 0 xcbc 12815 0 rmd160 16744 0 bnep 18281 2 rfcomm 47604 12 sha512_generic 12796 0 crypto_null 12918 0 parport_pc 32866 0 af_key 36389 0 ppdev 17113 0 binfmt_misc 17540 1 nfsd 281980 2 ext2 73795 1 nfs 436929 1 lockd 90326 2 nfsd,nfs fscache 61529 1 nfs auth_rpcgss 53380 2 nfsd,nfs nfs_acl 12883 2 nfsd,nfs sunrpc 255224 16 nfsd,nfs,lockd,auth_rpcgss,nfs_acl btusb 18332 2 vesafb 13844 2 pl2303 17957 1 ath3k 12961 0 bluetooth 180153 24 bnep,rfcomm,btusb,ath3k snd_hda_codec_hdmi 32474 4 nvidia 11308613 0 ftdi_sio 40679 1 usbserial 47113 6 pl2303,ftdi_sio psmouse 97485 0 snd_hda_codec_realtek 224173 1 snd_hda_intel 33719 5 snd_hda_codec 127706 3 snd_hda_codec_hdmi,snd_hda_codec_realtek,snd_hda_intel serio_raw 13211 0 snd_seq_midi 13324 0 snd_hwdep 17764 1 snd_hda_codec snd_pcm 97275 3 snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec snd_rawmidi 30748 1 snd_seq_midi snd_seq_midi_event 14899 1 snd_seq_midi snd_seq 61929 2 snd_seq_midi,snd_seq_midi_event snd_timer 29990 2 snd_pcm,snd_seq snd_seq_device 14540 3 snd_seq_midi,snd_rawmidi,snd_seq snd 79041 20 snd_hda_codec_hdmi,snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_rawmidi,snd_seq,snd_timer,snd_seq_device asus_atk0110 18078 0 mac_hid 13253 0 jc42 13948 0 soundcore 15091 1 snd snd_page_alloc 18529 2 snd_hda_intel,snd_pcm coretemp 13554 0 i2c_i801 17570 0 lp 17799 0 parport 46562 3 parport_pc,ppdev,lp r8169 62154 0 Any ideas? What does X do that's so important?

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  • How to make Bluetooth headset appear as a sound device?

    - by torbengb
    I have an onboard VIA sound card but no speakers attached. Instead, I want to use my bluetooth headset as my only sound device, both input and output (mono). I plugged in the bt dongle and was impressed that it was ready to use within seconds. I then paired it to my Jabra BT500 headset. No problems so far. I then went to Sound Preferences but no bluetooth is listed, only the onboard VIA sound card. Question: How can I enable my headset, and make it the default all the time?

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  • AS3 Font embedding problem in ComboBox on a loaded movie clip

    - by Arafat
    Hi all, I have three movie clips, say, LoaderMC, ChildMC1, and ChildMC2. ChildMC1 has TLFTextfields in it with embedded fonts. ChildMC2 has both TLFTextfields as well as combo boxes.(with embedded fonts) When I compile those movieclips separately, I can view the combo box texts without any problem. If I load the ChildMC1 and ChildMC2 into the LoaderMC, the texts doesn't appear at all on both TLFtextfields and Combo Boxes. I tried embedding the fonts in the LoaderMC, then, TLFTextfields was able to show the texts but still the ComboBoxes couldn't display the text. If I don't embed the fonts in the combo box, I can able to view the texts. What could be the problem? I read an article which says, "No one has completely understood the font embedding in flash AS3, we just have to do trial and error, to get it done" I don't know how far it is true, but in my case it seems, I should agree! Please help me...

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  • Using Delphi or FFMpeg to create a movie from image sequence

    - by Hein du Plessis
    Hi all My Delphi app has created a squence called frame_001.png to frame_100.png. I need that to be compiled into a movie clip. I think perhaps the easiest is to call ffmpeg from the command line, according to their documentation: For creating a video from many images: ffmpeg -f image2 -i foo-%03d.jpeg -r 12 -s WxH foo.avi The syntax foo-%03d.jpeg specifies to use a decimal number composed of three digits padded with zeroes to express the sequence number. It is the same syntax supported by the C printf function, but only formats accepting a normal integer are suitable. From: http://ffmpeg.org/ffmpeg-doc.html#SEC5 However my files are (lossless) png format, so I have to convert using imagemagick first. My command line is now: ffmpeg.exe -f image2 -i c:\temp\wentelreader\frame_%05d.jpg -r 12 foo.avi But then I get the error: [image2 @ 0x133a7d0]Could not find codec parameters (Video: mjpeg) c:\temp\wentelreader\Frame_C:\VID2EVA\Tools\Mencoder\wentel.bat5d.jpg: could not find codec parameters What am I doing wrong? Alternatively can this be done easily with Delphi?

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  • Detect movie being played (Windows)

    - by modosansreves
    Watching a movie is quite a different user activity. User doesn't touch neither mouse nor keyboard. Yet he 'actively' uses the computer. Thus, screensaver shouldn't run, indexing should be performed with care etc. On the other side, playing video requires either using direct write to video memory, or DirectShow, or some other API. This may be the key to the answer. What is the Dead Simple Way to determine that a video is being played?

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  • iOS 5 Audio Alarms Don't Sound Without kAudioSessionProperty_OverrideCategoryMixWithOthers On

    - by coneybeare
    I have an audio app that is having some problems with the way iOS 5 has changed audio behaviors. When my app's audio is playing (AVAudioSessionCategoryPlayback), and a Clock.app alarm or timer is fired from the OS, the UIAlertView notification pops up, but without the audio alert. My application sound ducks fine to get out of the way of the audio alert, but the alarm app's audio alert does not sound. Naturally, tons of support requests poured in over the iOS 5 change. I have solved this temporarily by setting kAudioSessionProperty_OverrideCategoryMixWithOthers which lets the alarm audio come through, but there are a few very undesirable side-effects when doing this: Other app's audio can play with/over mine. The remote control events are not routed to my app, but to iPod.app. None of the above drawbacks are acceptable for my app's requirements. I have been hacking away at this for some time now but haven't been able to crack it. How can I setup my audio such that: My app's audio still uses the AVAudioSessionCategoryPlayback category for background audio. The Clock.app alarms still have their audio alerts make sound The app still responds to remote control notifications

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  • Real Time Sound Captureing J2ME

    - by Abdul jalil
    i am capturing sound in J2me and send these bytes to remote system, i then play these bytes on remote system.five second voice is capture and send to remote system. i get the repeated sound again .i am making a sound messenger please help me where i am doing wrong i am using the follown code . String remoteTimeServerAddress="192.168.137.179"; sc = (SocketConnection) Connector.open("socket://"+remoteTimeServerAddress+":13"); p = Manager.createPlayer("capture://audio?encoding=pcm&rate=11025&bits=16&channels=1"); p.realize(); RecordControl rc = (RecordControl)p.getControl("RecordControl"); ByteArrayOutputStream output = new ByteArrayOutputStream(); OutputStream outstream =sc.openOutputStream(); rc.setRecordStream(output); rc.startRecord(); p.start(); int size=output.size(); int offset=0; while(true) { Thread.currentThread().sleep(5000); rc.commit(); output.flush(); size=output.size(); if(size0) { recordedSoundArray=output.toByteArray(); outstream.write(recordedSoundArray,0,size); } output.reset(); rc.reset(); rc.setRecordStream(output); rc.startRecord(); }

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  • T60 Screen/LCD gets black after some minutes with a highpitched sound rising and fading

    - by Edward De Leau
    Just now my T60 screen got "black" (so no display). On my second monitor: no problems so the vga output works. symptom: Screen blanks / no display but works on second monitor steps to reproduce: - boot - wait (it does not matter what you do you do not have to login or anything) - (now the monitor of the laptop slowly begins to make a ssssssssHHHHHHHHHHHHHHHHHWOEOEssssssss noice of about 10 seconds) - right after the sounds ends the monitor gets black times seem to be the same each time. software: installed no new software before/after, running zone alarm and antivirus. other: it does not feel hot in any place, there dont seem to be running processes with strange behaviour. warranty: out of warranty what was i doing: typing text on a website and doing some php coding in a text editor. Anyone any idea what I can do here other than buy a new laptop? / does it sound familiar to known cases? update: * exactly the same problem: http://forums.lenovo.com/t5/T61-and-prior-T-series-ThinkPad/T60-Screen-Blackout/m-p/288772 and the second poster (garyj) here: http://forums.lenovo.com/t5/T61-and-prior-T-series-ThinkPad/Black-Screen-on-T60/m-p/235053#M48627 and here: "i have that same problem. i replaced the CCRL on mine and it works fine when the screen is not screwed in. once the frame of the LCD screen (metal portion) touches the metal on the laptop which holds the screen the screen goes black. If the metal is touching the screen when you boot up it boots up with it being very dimmly lit. " from http://forums.lenovo.com/t5/T61-and-prior-T-series-ThinkPad/T60-screen-problems/m-p/205047#M44995 (it seems replacing the lcd is no use, he tried it 3 times). same problem: http://forums.lenovo.com/t5/T61-and-prior-T-series-ThinkPad/T60-black-screen/m-p/80604#M25914 Hmmm... not handy 3 or 4 months ago I ordered and installed a new fan. Now the LCD. Which does not seem the core isse but some electric issue so it seems replacing the LCD is not the thing todo here. I have the following question: If it is not the LCD that needs to be replaced (see other threads) which parts can I order to fix this? Has anyone got any information which could lead me to identify the issue? I have read replacing the "inverter" AND the "backlightning" would that make sense?

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  • jQuery DOM element creation vs innerHTML

    - by Mohammad
    While having one of my questions answered, cletus mentioned that when creating elements in jQuery it's better to use direct DOM element creation, instead of innerHTML. I tried googling it but I wasn't able to find a good article with comparisons. I've provided this code bellow as an example and I was wondering if someone could help me rewrite it in direct DOM element creation form in hope that i would also learn the difference afterwards. var img = $(this); img.append('<p class="cap"><a href="'+img.parent().attr('href')+'">'+ img.attr('title')+'</p></a>'); Thanks so much.

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  • How to play small sound file continuously in Silverlight?

    - by ash
    Hello, I have two questions regarding Silverlight's SoundPlay action and properties. My scenario is like: I have two story board: The first story board has an image and a sound file; when the silverlight application gets loaded, the sound starts to play automatically, but if someone clicks the image, the sound file will stop and the second storyboard will start with a new sound file. 1) My first question is how to stop the first sound file of first story board when the second story board starts with the second sound file. 2) My second question is how to play a sound file continuously; for example, in Silverlight we can play a story board continuously with RepeatBehavior="Forever"; but I cannot find a way to play my 10 second sound file forever or continuously. Note: I have attached a small XAML file to show what I am talking about; I am also stating that if instead of an image file, if there were a button, then I can stop the first music file after I click the button and start my second story board with a new sound file, but I would like to use image file instead of a button. Is it possible? If it is, how to do it? Therefore, please answer my following two questions or give big hint or website tutorial links on 1) How to stop the first sound file of first story board when the second story board starts with the second sound file ( When the clickable element is an image instead of a button) 2) How to play a 10 second sound file continuously? ............Code Snippet...................... XAML ............ <Grid x:Name="LayoutRoot" Background="Red"> <Button HorizontalAlignment="Left" Margin="212,0,0,111" VerticalAlignment="Bottom" Width="75" Content="Button" Click="onClick"/> <MediaElement x:Name="sound2_mp3" Height="0" HorizontalAlignment="Left" Margin="105,230,0,0" VerticalAlignment="Top" Width="0" Source="/sound2.mp3" Stretch="Fill"/> <MediaElement x:Name="sound1_mp1" Height="0" HorizontalAlignment="Left" Margin="190,164,0,0" VerticalAlignment="Top" Width="0" Source="/sound1.mp3" Stretch="Fill" AutoPlay="False"/> </Grid> ..................................................................................................................................................................................................................... using System; using System.Windows; using System.Windows.Controls; using System.Windows.Documents; using System.Windows.Ink; using System.Windows.Input; using System.Windows.Media; using System.Windows.Media.Animation; using System.Windows.Shapes; namespace testPrj { public partial class MainPage : UserControl { public MainPage() { // Required to initialize variables InitializeComponent(); } private void onClick(object sender, System.Windows.RoutedEventArgs e) { Storyboard1.Stop(); sound2_mp3.Stop(); sound1_mp1.Play(); } } } ...................................................................................................

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  • How to play audio in Java Application

    - by user577829
    I'm making a java application and I need to play audio. I'm playing mainly small sound files of my cannon firing (its a cannon shooting game) and the projectiles exploding, though I plan on having looping background music. I have found two different methods to accomplish this, but both don't work how I want. The first method is literally a method: public void playSoundFile(File file) {//http://java.ittoolbox.com/groups/technical-functional/java-l/sound-in-an-application-90681 try { //get an AudioInputStream AudioInputStream ais = AudioSystem.getAudioInputStream(file); //get the AudioFormat for the AudioInputStream AudioFormat audioformat = ais.getFormat(); System.out.println("Format: " + audioformat.toString()); System.out.println("Encoding: " + audioformat.getEncoding()); System.out.println("SampleRate:" + audioformat.getSampleRate()); System.out.println("SampleSizeInBits: " + audioformat.getSampleSizeInBits()); System.out.println("Channels: " + audioformat.getChannels()); System.out.println("FrameSize: " + audioformat.getFrameSize()); System.out.println("FrameRate: " + audioformat.getFrameRate()); System.out.println("BigEndian: " + audioformat.isBigEndian()); //ULAW format to PCM format conversion if ((audioformat.getEncoding() == AudioFormat.Encoding.ULAW) || (audioformat.getEncoding() == AudioFormat.Encoding.ALAW)) { AudioFormat newformat = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED, audioformat.getSampleRate(), audioformat.getSampleSizeInBits() * 2, audioformat.getChannels(), audioformat.getFrameSize() * 2, audioformat.getFrameRate(), true); ais = AudioSystem.getAudioInputStream(newformat, ais); audioformat = newformat; } //checking for a supported output line DataLine.Info datalineinfo = new DataLine.Info(SourceDataLine.class, audioformat); if (!AudioSystem.isLineSupported(datalineinfo)) { //System.out.println("Line matching " + datalineinfo + " is not supported."); } else { //System.out.println("Line matching " + datalineinfo + " is supported."); //opening the sound output line SourceDataLine sourcedataline = (SourceDataLine) AudioSystem.getLine(datalineinfo); sourcedataline.open(audioformat); sourcedataline.start(); //Copy data from the input stream to the output data line int framesizeinbytes = audioformat.getFrameSize(); int bufferlengthinframes = sourcedataline.getBufferSize() / 8; int bufferlengthinbytes = bufferlengthinframes * framesizeinbytes; byte[] sounddata = new byte[bufferlengthinbytes]; int numberofbytesread = 0; while ((numberofbytesread = ais.read(sounddata)) != -1) { int numberofbytesremaining = numberofbytesread; sourcedataline.write(sounddata, 0, numberofbytesread); } } } catch (Exception e) { e.printStackTrace(); } } The problem with this is that my entire program stops until the sound file is finished, or at least nearly finished. The second method is this: File file = new File("Launch1.wav"); AudioClip clip; try { clip = JApplet.newAudioClip(file.toURL()); clip.play(); } catch (Exception e) { e.getMessage(); } The problem I have here is that every time the sound file ends early or doesn't play at all depending on where I place the code. Is their any way to play sound without the above mentioned problems? Am I doing something wrong? Any help is greatly appreciated.

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  • Understanding flow of object creation

    - by Boris Mandovskes
    I'm new to java and I wonder if there is simple way to know flow like the following of object creation, I'm using eclipse and when I write new ObjectInputStream and press CTRL+SPACE. I don't see any option that I can enter new BufferedInputStream (I have copied the code from example) and than to create new object for FileInputStream etc. in = new ObjectInputStream(new BufferedInputStream(new FileInputStream("emp.dat"))); List temp = (List)in.readObject(); I give that example since this is the first time that I saw this kind of creation new object flow and I want to use some best practice for the next times.

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  • Profile creation process stuck halfway Websphere

    - by ngubk
    I'm creating a cell profile on Linux Mint 12 , WAS 8.0 Network Deployment Trial. But using manageProfiles.sh or Profile Management Tool, I can not create any profile (cell, application ...). When I check the log file, the profile creation process is always stop halfway (does not show any error, just stuck there). The log is always like this <record> <date>2012-11-02T04:11:15</date> <millis>1351847475108</millis> <sequence>2985</sequence> <logger>com.ibm.ws.install.configmanager.actionengine.ant.utils.ANTLogToCmtLogAdapter</logger> <level>INFO</level> <class>com.ibm.ws.install.configmanager.actionengine.ant.utils.ANTLogToCmtLogAdapter</class> <method>messageLogged</method> <thread>0</thread> <message>replacing value for user.install.root (null) with (/opt/IBM/WebSphere/AppServer/profiles)</message> </record> <record> <date>2012-11-02T04:11:15</date> <millis>1351847475108</millis> <sequence>2986</sequence> <logger>com.ibm.ws.install.configmanager.actionengine.ant.utils.ANTLogToCmtLogAdapter</logger> <level>INFO</level> <class>com.ibm.ws.install.configmanager.actionengine.ant.utils.ANTLogToCmtLogAdapter</class> <method>messageLogged</method> <thread>0</thread> <message>replacing value for was.install.root (/opt/IBM/WebSphere/AppServer) with (/opt/IBM/WebSphere/AppServer)</message> </record> <record> <date>2012-11-02T04:11:15</date> <millis>1351847475108</millis> <sequence>2987</sequence> <logger>com.ibm.ws.install.configmanager.actionengine.ant.utils.ANTLogToCmtLogAdapter</logger> <level>INFO</level> <class>com.ibm.ws.install.configmanager.actionengine.ant.utils.ANTLogToCmtLogAdapter</class> <method>messageLogged</method> <thread>0</thread> <message>replacing value for was.repository.root (null) with (/opt/IBM/WebSphere/AppServer/profiles/config)</message> </record> <record> <date>2012-11-02T04:11:15</date> <millis>1351847475108</millis> <sequence>2988</sequence> <logger>com.ibm.ws.install.configmanager.actionengine.ant.utils.ANTLogToCmtLogAdapter</logger> <level>INFO</level> <class>com.ibm.ws.install.configmanager.actionengine.ant.utils.ANTLogToCmtLogAdapter</class> <method>messageLogged</method> <thread>0</thread> <message>replacing value for com.ibm.ws.scripting.wsadminprops (null) with (/opt/IBM/WebSphere/AppServer/profiles/properties/wsadmin.properties)</message> </record> <record> <date>2012-11-02T04:11:15</date> <millis>1351847475120</millis> <sequence>2989</sequence> <logger>com.ibm.ws.install.configmanager.actionengine.ant.utils.ANTLogToCmtLogAdapter</logger> <level>INFO</level> <class>com.ibm.ws.install.configmanager.actionengine.ant.utils.ANTLogToCmtLogAdapter</class> <method>messageLogged</method> <thread>0</thread> <message>Resetting listener available status to: false</message> </record> <record> <date>2012-11-02T04:11:15</date> <millis>1351847475121</millis> <sequence>2990</sequence> <logger>com.ibm.ws.install.configmanager.actionengine.ant.utils.ANTLogToCmtLogAdapter</logger> <level>INFO</level> <class>com.ibm.ws.install.configmanager.actionengine.ant.utils.ANTLogToCmtLogAdapter</class> <method>messageLogged</method> <thread>0</thread> <message>setting wsadmin requester timeouts</message> </record> <record> <date>2012-11-02T04:11:15</date> <millis>1351847475128</millis> <sequence>2991</sequence> <logger>com.ibm.ws.install.configmanager.actionengine.ant.utils.ANTLogToCmtLogAdapter</logger> <level>INFO</level> <class>com.ibm.ws.install.configmanager.actionengine.ant.utils.ANTLogToCmtLogAdapter</class> <method>messageLogged</method> <thread>0</thread> <message>wsadmin requester retry count = 240000, initialization retry count = 12000, shutdown retry count = 12000</message> </record> <record> <date>2012-11-02T04:11:15</date> <millis>1351847475128</millis> <sequence>2992</sequence> <logger>com.ibm.ws.install.configmanager.actionengine.ant.utils.ANTLogToCmtLogAdapter</logger> <level>INFO</level> <class>com.ibm.ws.install.configmanager.actionengine.ant.utils.ANTLogToCmtLogAdapter</class> <method>messageLogged</method> <thread>0</thread> <message>Checking for wsadmin listener initialization</message> </record>

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