Search Results

Search found 12365 results on 495 pages for 'core audio'.

Page 11/495 | < Previous Page | 7 8 9 10 11 12 13 14 15 16 17 18  | Next Page >

• Tool to bulk speed up/convert an audio file

  - by User1

  I want to listen to certain podcasts on my phone but I have two common problems: The audio is in some weird format (some don't play on my phone). The audio is slow. I want to use something like sox or avconv to bulk convert the files. Since this is just voice and going on a cell phone, small low-quality files would be best for me. I had some good success using avconv: avconv -i weird.wma normal.ogg Unforunately, this command creates an enormous ogg file and I can't get it play faster. Ideally, this particular file would play at 170% of the original speed.

  Read the article

• Solutions for exporting a remote desktop app (display and audio)

  - by Richard

  I'm looking for a solution that will allow me to export a desktop app running on a server to a client machine. The server is ideally Linux, the desktop is Windows (+Mac for icing on the cake). The export should be encrypted and I need to support multiple clients from one server. I only want to export an individual app, not a whole desktop, and ideally am looking for open source solutions. The obvious, cheapest, simplest choice is to use X tunnelled over ssh (e.g using Xming on the desktop) but X doesn't support audio. What are the alternatives? Or is there a way to support audio using X or in parallel to X? Thanks

  Read the article

• FreePBX: Asterisk in the Cloud (EC2) Audio Problems

  - by neezer

  Please pardon the newbie question, but I can't seem to figure this out. I followed the Voxilla's tut to the tee: http://voxilla.com/2009/10/15/voxill...p-by-step-1457 But in making calls, my softphones connect, yet no audio (in either direction). I know from poking around the forums that this is generally caused by two factors: NAT and audio codecs. I (being new to the arena), however, don't know which. I believe I have Asterisk and the clients restricted to just ulaw, and I also believe I have the correct ports open, and my externip set correctly (I think the Voxilla AMI does this automatically, since it's in the cloud). I'm a bit lost. I'd be happy to post whatever configuration files that might help, provided you tell me where they are on the filesystem. But like I said before, this is effectively a vanilla install of Voxilla's own FreePBX AMI. I'd appreciate any help or guidance here. Thanks!

  Read the article

• FreePBX: Asterisk in the Cloud (EC2) Audio Problems

  - by neezer

  Please pardon the newbie question, but I can't seem to figure this out. I followed the Voxilla's tut to the tee: http://voxilla.com/2009/10/15/voxill...p-by-step-1457 But in making calls, my softphones connect, yet no audio (in either direction). I know from poking around the forums that this is generally caused by two factors: NAT and audio codecs. I (being new to the arena), however, don't know which. I believe I have Asterisk and the clients restricted to just ulaw, and I also believe I have the correct ports open, and my externip set correctly (I think the Voxilla AMI does this automatically, since it's in the cloud). I'm a bit lost. I'd be happy to post whatever configuration files that might help, provided you tell me where they are on the filesystem. But like I said before, this is effectively a vanilla install of Voxilla's own FreePBX AMI. I'd appreciate any help or guidance here. Thanks!

  Read the article

• Audio input problem in Ubuntu 9.10

  - by Andrea Ambu

  My audio input is a mix of my mic output and my sound card output. I'd like it to be just my mic output. I was able to do so in Ubuntu 9.04 but the interface is 9.10 is totally changed and I tried every my creativity was able to think. It's really annoying when talking to other people over the internet because they keep hearing their voice back. I'm not sure I explained it in clear way so I'll give you an example: What I do: I put an mp3 on play or a video on youtube then open a recorder and start to talk on my mic. What happens: both my voice and audio from mp3/youtube get reordered, even if I put headphones volume to 0 (via hardware). What I'd like to happen: Only my voice should be recorded. I'm sure I'm missing some technical term, but that's the problem and I'd like to solve it in Ubuntu 9.10, any idea?

  Read the article

• Screenflick Audio option in MacBook Pro

  - by John

  When after I shut down my MacBook Pro by holding the power button for a few sec, (which I found is bad for the computer, so I will not do anymore) I found that my speaker doesn't play until I plug in and out earphone into the machine. When my speaker is not working like this, and when I am on a random webcam chatting site like chatroulette.com, they can hear the music playing on my iTunes when I choose Screenflick Audio option in the Mic setting. But when the Speaker is working back again, they don't hear the music playing even when I do Screenflick Audio mode. How can I make it work? Also, how do you make the chatting partner hear my music playing on my computer while I talk to them (not via my speaker, since it's bad quality).

  Read the article

• Audio problems with asus notebook with Bluetooth and usb devices in win 7

  - by QuickSilver

  My notebook is Asus P53E - core i5 Windows 7 installed Audio from PC speakers and headphone is distorted when i turn on bluetooth or some usb device plugged in. I belive this is a software issue. I tried updating my audio drivers but nothing help. Any help will be appreciated. Update: After a few days digging I found that this problem is causing by the asus sound enhancement application SonicFocus. The distortion stops while turning off sonic focus. Can anyone help me with a solution other than turning off SonicFocus

  Read the article

• Windows audio service fails to automatically start after VirtualBox install

  - by humble_coder

  I'm having a completely nonsensical issue in Windows XP SP3. Basically my "Windows Audio" service no longer starts automatically. Despite being set to "Automatic" I have to manually go in and start it. This issue didn't start until the most recent update of VirtualBox, but I can't find anything on the forums related to this specific issue. I've tried reinstalling the RealTek drivers as well, in the event that that had something to do with it. Any assistance is most appreciated! EDIT 1: It is the host's audio that won't start. The update of Virtualbox was merely the "marker" of when these events started occurring. Given it's the only variable/change I'm assuming a correlation.

  Read the article

• No Audio Output Device is installed

  - by mabho

  Hi, this is an intermitent problem in my Sony Vaio model PCG-5K1L. I keep on getting a "No Audio Output Device is installed" when hovering my loudspeaker icon in Windows Vista. I have tried System Device Manager Sound Realtek High Definition Album Update Driver Software. The update process went through, but nothing happens. Still Vista does not seem to recognize my audio software. The strange part is that out of nothing my sound card can resume working to stop again hours later... If someone has any clues to solve this, please, help. Thanks a lot.

  Read the article

• Redirect audio from laptop to desktop over LAN

  - by Ram Rachum

  I want to be able to play a song on my laptop and have it sound through my desktop's (infinitely better) speakers. If you're familiar with Input Director: I want something that is to audio what Input Director is to mouse/keyboard. I want something that automatically redirects all audio from the laptop to the desktop in real time, and I want that solution to require, like Input Director, minimum maintenance. Beyond the initial setup, I don't want to have to babysit the program that does this. I want something that launches automatically with Windows and just works, and also allows me to cancel it whenever I want. And also doesn't go crazy when the laptop is turned on in a different network where the desktop computer isn't available. Any suggestions for such a program? (I use Windows XP on both computers.)

  Read the article

• How can I select an audio output device in directshow

  - by Vibhore Tanwer

  I was wondering how I can select the output device for audio in directshow. I am able to get available audio output devices in directshow. But how can I make one of these to be audio output device. Its always going for the default audio device. I want to be able to output audio on my choice of device. I have been struggling through google but couldn't find anything useful. All I could get was this link but it doesn't really solve my problem. Any help will be really helpful for me.

  Read the article

• Read the audio input level peak in Cocoa

  - by Kenneth Ballenegger

  I'm trying to make an audio-sensitive animation, and for that purpose, I'm looking for a way to look up the current audio level. I'm looking for the peak within a set amount of time. (Think the red bar that stays on for a second or so, on an audio meter.) I've searched around for for something like this, and the only thing I could find was how to read a movie's audio levels, and how Quartz Compositions have access to this thru their iTunes Visualizer protocol. I'm looking for a way to read this from the microphone, although I'm also interested if you know how to read this from an audio file. Thanks!

  Read the article

• Chrome/Webkit audio tag bug?

  - by Ronald

  I'm trying to get HTML5's audio tag to work in Chrome. The following code works flawlessly in Firefox, any ideas why it isn't working in Webkit? <html> <head> <script type="text/javascript"> function init(){ audio = new Audio("chat.ogg"); audio.play(); } </script> </head> <body onload="init()"> </body> I should also note that I tried this with an mp3 as well. Regardless of what format, whenever .play() is called on audio, Chrome responds with "undefined".

  Read the article

• Traktor Audio 2 DJ soundcard configuration

  - by Jaroslav

  I have a Traktor Audio 2 DJ USB sound card (the first version of what it's now called simply Traktor Audio 2) The problem in settings it only sees one output, when there should be two (I need that for Mixxx etc.) Also I want to be able set the sample rate to one of these: 44.1, 48, 88.2, 96 kHz or at least check which one is set. Additionally if possible setting the latency would be an advantage. Some info: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: TraktorAudio2 [Traktor Audio 2], device 0: Traktor Audio 2 [Traktor Audio 2] Subdevices: 1/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 $ cat /proc/asound/cards 0 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xfdcfc000 irq 45 1 [TraktorAudio2 ]: snd-usb-caiaq - Traktor Audio 2 Native Instruments Traktor Audio 2 (usb-0000:00:1d.7-8)

  Read the article

• Core data migration failing with "Can't find model for source store" but managedObjectModel for source is present

  - by Ira Cooke

  I have a cocoa application using core-data, which is now at the 4th version of its managed object model. My managed object model contains abstract entities but so far I have managed to get migration working by creating appropriate mapping models and creating my persistent store using addPersistentStoreWithType:configuration:options:error and with the NSMigratePersistentStoresAutomaticallyOption set to YES. NSDictionary *optionsDictionary = [NSDictionary dictionaryWithObject:[NSNumber numberWithBool:YES] forKey:NSMigratePersistentStoresAutomaticallyOption]; NSURL *url = [NSURL fileURLWithPath: [applicationSupportFolder stringByAppendingPathComponent: @"MyApp.xml"]]; NSError *error=nil; [theCoordinator addPersistentStoreWithType:NSXMLStoreType configuration:nil URL:url options:optionsDictionary error:&error] This works fine when I migrate from model version 3 to 4, which is a migration that involves adding attributes to several entities. Now when I try to add a new model version (version 5), the call to addPersistentStoreWithType returns nil and the error remains empty. The migration from 4 to 5 involves adding a single attribute. I am struggling to debug the problem and have checked all the following; The source database is in fact at version 4 and the persistentStoreCoordinator's managed object model is at version 5. The 4-5 mapping model as well as managed object models for versions 4 and 5 are present in the resources folder of my built application. I've tried various model upgrade paths. Strangely I find that upgrading from an early version 3 - 5 works .. but upgrading from 4 - 5 fails. I've tried adding a custom entity migration policy for migration of the entity whose attributes are changing ... in this case I overrode the method beginEntityMapping:manager:error: . Interestingly this method does get called when migration works (ie when I migrate from 3 to 4, or from 3 to 5 ), but it does not get called in the case that fails ( 4 to 5 ). I'm pretty much at a loss as to where to proceed. Any ideas to help debug this problem would be much appreciated.

  Read the article

• moving audio over a local network using GStreamer

  - by James Turner

  I need to move realtime audio between two Linux machines, which are both running custom software (of mine) which builds on top of Gstreamer. (The software already has other communication between the machines, over a separate TCP-based protocol - I mention this in case having reliable out-of-band data makes a difference to the solution). The audio input will be a microphone / line-in on the sending machine, and normal audio output as the sink on the destination; alsasrc and alsasink are the most likely, though for testing I have been using the audiotestsrc instead of a real microphone. GStreamer offers a multitude of ways to move data round over networks - RTP, RTSP, GDP payloading, UDP and TCP servers, clients and sockets, and so on. There's also many examples on the web of streaming both audio and video - but none of them seem to work for me, in practice; either the destination pipeline fails to negotiate caps, or I hear a single packet and then the pipeline stalls, or the destination pipeline bails out immediately with no data available. In all cases, I'm testing on the command-line just gst-launch. No compression of the audio data is required - raw audio, or trivial WAV, uLaw or aLaw encoding is fine; what's more important is low-ish latency.

  Read the article

• Audio output from Silverlight

  - by leecarter

  I'm looking to develop a Silverlight application which will take a stream of data (not an audio stream as such) from a web server. The data stream would then be manipulated to give audio of a certain format (G.711 a-Law for example) which would then be converted into PCM so that additional effects can be applied (such as boosting the volume). I'm OK up to this point. I've got my data, converted the G.711 into PCM but my problem is being able to output this PCM audio to the sound card. I basing a solution on some C# code intended for a .Net application but in Silverlight there is a problem with trying to take a copy of a delegate (function pointer) which will be the topic of a separate question once I've produced a simple code sample. So, the question is... How can I output the PCM audio that I have held in a data structure (currently an array) in my Silverlight to the user? (Please don't say write the byte values to a text box) If it were a MP3 or WMA file I would play it using a MediaElement but I don't want to have to make it into a file as this would put a crimp on applying dynamic effects to the audio. I've seen a few posts from people saying low level audio support is poor/non-existant in Silverlight so I'm open to any suggestions/ideas people may have.

  Read the article

• RDS installation failure on 2012 R2 Server Core VM in Hyper-V Server

  - by Giles

  I'm currently installing a test-bed for my firms Infrastructure replacement. 10 or so Windows/Linux servers will be replaced by 2 physical servers running Hyper-V server. All services (DC, RDS, SQL) will be on Windows 2012 R2 Server Core VMs, Exchange on Server 2012 R2 GUI, and the rest are things like Elastix, MailArchiver etc, which aren't part of the equation thus far. I have installed Hyper-V server on a test box, and sucessfully got two virtual DC's running, SQL 2014 running, and 8.1 which I use for the RSAT tools. When trying to install RDS (The old fashioned kind, not the newer VDI(?) style), I get a failed installation due to the server not being able to reboot. A couple of articles have said not to do it locally, so I've moved on. Sitting at the Powershell prompt on the Domain Controller or SQL server (Both Server Core), I run the following commands: Import-Module RemoteDesktop New-SessionDeployment -ConnectionBroker "AlstersTS.Alsters.local" -SessionHost "AlstersTS.Alsters.local" The installation begins, carries on for 2 or 3 minutes, then I receive the following error message: New-SessionDeployment : Validation failed for the "RD Connection Broker" parameter. AlstersTS.Alsters.local Unable to connect to the server by using WindowsPowerShell remoting. Verify that you can connect to the server. At line:1 char:1 + NewSessionDeployment -ConnectionBroker "AlstersTS.Alsters.local" -SessionHost " ... + + CategoryInfo : NotSpecified: (:) [Write-Error], WriteErrorException + FullyQualifiedErrorID : Microsoft.PowerShell.Commands.WriteErrorException,New-SessionDeployment So far, I have: Triple, triple checked syntax. Tried various other commands, and a script to accomplish the same task. Checked DNS is functioning as it should. Checked to the best of my knowledge that AD is working as it should. Checked that the Network Service has the needed permissions. Created another VM and placed the two roles on different servers. Deleted all VMs, started again with a new domain name (Lather, rinse, repeat) Performed the whole installation on a second physical box running Hyper-V Server Pleaded with it Interestingly, if I perform the installation via a GUI installation, the thing just works! Now I know I could convert this to a Server Core role after installation, but this wouldn't teach me what was wrong in the first instance. I've probably got 10 pages through various Google searches, each page getting a little less relevant. The closest matches seem to have good information, but it doesn't seem to be the fix for my set-up. As a side note, I expected to be able to "tee" or "out-file" the error message into a text file, but couldn't get that to work either, so I've typed in the error message manually. Chaps, any suggestions, from the glaringly obvious, to the long-winded and complex? Thanks!

  Read the article

• Windows Web Server 2008 R2 Server Core local password complexity

  - by Dennis Allen

  How can I disable the local user account password complexity settings on Windows 2008 R2 "Server Core"? I am trying to migrate our windows 2003 web server to windows 2008 R2. I am trying to see if I can use the "Server Core" install, and it has been a very internet search intensive experience. What I can't find out how to do is to find out how to disable password complexity for local user accounts. While our user account generator currently creates nice strong passwords, there was a time when this was not the case and unfortunately forcing the users to change their password is not an option at this time. Any help greatly appreciated. Dennis

  Read the article

• Crash in audio resampler with some audio rates - FFMPEG PHP ( Solved! )

  - by Olaf Erlandsen

  i have a problem with this command( FFMPEG PHP ): Command: ffmpeg -i 62f76f050494f0ed6a5997967c00c0c0.wmv -ss 0 -t 99 -y -ar 44100 -async 44100 -r 29.970 -ac 2 -qscale 5 -f flv 62f76f050494f0ed6a5997967c00c0c0.flv Output: FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers built on Jan 29 2012 17:52:15 with gcc 4.4.5 20110214 (Red Hat 4.4.5-6) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --enable-avfilter --enable-avfilter-lavf --enable-libdc1394 --enable-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab libavutil 50.15. 1 / 50.15. 1 libavcodec 52.72. 2 / 52.72. 2 libavformat 52.64. 2 / 52.64. 2 libavdevice 52. 2. 0 / 52. 2. 0 libavfilter 1.19. 0 / 1.19. 0 libswscale 0.11. 0 / 0.11. 0 libpostproc 51. 2. 0 / 51. 2. 0 [asf @ 0xe81670]max_analyze_duration reached Input #0, asf, from '/var/www/resources/tmp/62f76f050494f0ed6a5997967c00c0c0.wmv': Metadata: WMFSDKVersion : 12.0.7601.17514 WMFSDKNeeded : 0.0.0.0000 IsVBR : 0 Duration: 00:00:50.87, bitrate: 2467 kb/s Stream #0.0: Audio: wmapro, 44100 Hz, stereo, flt, 256 kb/s Stream #0.1: Video: vc1, yuv420p, 950x460 [PAR 1:1 DAR 95:46], 25 fps, 25 tbr, 1k tbn, 25 tbc Output #0, flv, to '/var/www/resources/media/62f76f050494f0ed6a5997967c00c0c0.flv': Metadata: encoder : Lavf52.64.2 Stream #0.0: Video: flv, yuv420p, 950x460 [PAR 1:1 DAR 95:46], q=2-31, 200 kb/s, 1k tbn, 29.97 tbc Stream #0.1: Audio: libmp3lame, 11025 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #0.1 Press [q] to stop encoding frame= 72 fps= 0 q=5.0 size= 0kB time=10.91 bitrate= 0.0kbits/s Multiple frames in a packet from stream 0 Warning, using s16 intermediate sample format for resampling frame= 141 fps=139 q=5.0 size= 103kB time=8.15 bitrate= 103.2kbits/s frame= 220 fps=144 q=5.0 size= 875kB time=10.92 bitrate= 656.6kbits/s frame= 290 fps=143 q=5.0 size= 1525kB time=13.74 bitrate= 909.1kbits/s frame= 356 fps=141 q=5.0 size= 2153kB time=15.99 bitrate=1103.1kbits/s frame= 427 fps=141 q=5.0 size= 2847kB time=18.70 bitrate=1247.0kbits/s frame= 497 fps=141 q=5.0 size= 3771kB time=21.16 bitrate=1460.0kbits/s frame= 575 fps=142 q=5.0 size= 4695kB time=24.61 bitrate=1563.0kbits/s frame= 639 fps=141 q=5.0 size= 5301kB time=26.80 bitrate=1620.2kbits/s frame= 703 fps=139 q=5.0 size= 5829kB time=29.36 bitrate=1626.2kbits/s frame= 774 fps=139 q=5.0 size= 6659kB time=32.39 bitrate=1684.0kbits/s frame= 842 fps=139 q=5.0 size= 7915kB time=35.27 bitrate=1838.6kbits/s frame= 911 fps=139 q=5.0 size= 9011kB time=37.98 bitrate=1943.4kbits/s frame= 975 fps=138 q=5.0 size= 9788kB time=40.59 bitrate=1975.3kbits/s frame= 1041 fps=138 q=5.0 size= 10904kB time=43.83 bitrate=2037.9kbits/s frame= 1115 fps=138 q=5.0 size= 11795kB time=46.24 bitrate=2089.8kbits/s frame= 1183 fps=138 q=5.0 size= 12678kB time=48.74 bitrate=2130.7kbits/s frame= 1247 fps=137 q=5.0 size= 13964kB time=51.36 bitrate=2227.5kbits/s frame= 1271 fps=136 q=5.0 Lsize= 15865kB time=58.86 bitrate=2208.1kbits/s video:15366kB audio:462kB global headers:0kB muxing overhead 0.238956% Problem: Warning, using s16 intermediate sample format for resampling I've also tried changing the parameter From -ar 44100 to -ar 11025 Thanks! Solution: Read this link: http://en.wikipedia.org/wiki/MP3#Bit_rate

  Read the article

• Windows Server 2008 R2 Server Core with AD Role having GUI Admin Console

  - by Robert Koritnik

  I would like to setup a machine with Windows Server 2008 R2 Server Core and install following server roles: Active Directory Domain Services Active Directory Federation Services Active Directory Lightweight Directory Services (I'm not sure whether I actually need this one - see note below) I'm obviously going to install Enterprise Edition. Question Can I have an AD administration graphical user interface to manage Active Directory on Server Core machine? I would really like to have it, because I'm not so keen to do stuff using power-shell, because I've never managed AD as well, so a GUI would be much more helpful, because I could at least visualize it a bit better and maybe understand AD structures. Note: I'm setting up development environment machine as well and installing Sharepoint Foundation 2010 on in so it would use this AD machine.

  Read the article

• Will Intel be releasing anymore 6-core processors soon?

  - by jasondavis

  I am about to start buying parts every week for as long as it takes me to build the best PC I can build. I am looking at the Intel i7-920 processor right now because it is about 250$ and it is a quad-core processor based on the x58 chipset I believe. From what I have read so far, intel is coming out with some 6-core processors soon that will also use the x58 chipset and will allow me to use the same motherboard and memory/ram to upgrade to a 6-core. This sounds really good to me right now. I just read that the new 6-core processor. The Core i7-980X (extreme edition) was just released which is the first 6-core processor but it is supposed to be around $1,000 so I will probably just get the i7-920 for now and then upgrade to the 6-core version when the price goes down. The motherboard I am looking at getting the GIGABYTE GA-X58A-UD5 which is around $280 at newegg.com So that is my basic plan SO far. I have not purchased any parts yet. I am just wanting to ask if this sounds like a good idea or if I should wait longer if I am wanting to eventually have a 6-core processor. Does anyone know if Intel is planning on releasing any other 6-core processor in addition to the Core i7-980X in the near future? I just want to make sure I am buying the best setup for my money if I am going all out on it, thanks for any tips/advice.

  Read the article

• Dual Core or Quad Core CPU for NetBeans/Eclipse development?

  - by cdb

  I am going to buy a new desktop CPU. I am a programmer who mainly uses NetBeans IDE for Java web application development, with GlassFish application server. I went through the discussion regarding Dual Core or Quad Core. My doubt is that software like IDEs (NetBeans, Eclipse, etc., with a server running) may not be written with multiple cores in mind? I am not a game addict... So what is best for me, and which company should I choose, AMD/Intel?

  Read the article

• JSP / Tomcat / Apache setup overview on Fedora Core

  - by Richard T

  Hi Folks, For someone with so much Java experience, boy do I feel clueless - thanks in advance for your help in my grocking the present (Feb, 2010) JSP environment. Here's what I am hoping to learn: Do I understand correctly that most people use Apache to "front-end" their Tomcat servers, such that Apache "talks" directly to web clients and "proxies" Tomcat servers? Do I understand correctly that Apache isn't capable of serving JSP directly but requires a server (like Tomcat)? Is there an RPM package for Fedora Core so I don't have to build one myself? Or, does Fedora Core's package installer do a good job on this one from source code? (Some do, some don't!) While I'm here asking questions; Does Tomcat come with a working example that one can start hacking on as a way to get started quickly? If not, got a good suggestion? Thanks folks, RT

  Read the article

• Any way for ubuntu to use more than one core of i7 cpu on my Asus laptop?

  - by G. He

  Newly installed ubuntu 11.10 on a new Asus U46E laptop. /proc/cpuinfo correctly identified the cpu but shows only one core: processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 42 model name : Intel(R) Core(TM) i7-2640M CPU @ 2.80GHz stepping : 7 cpu MHz : 800.000 cache size : 4096 KB physical id : 0 siblings : 1 core id : 0 cpu cores : 1 apicid : 0 initial apicid : 0 fpu : yes fpu_exception : yes cpuid level : 13 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe syscall nx rdtscp lm constant_tsc up arch_perfmon pebs bts rep_good nopl xtopology nonstop_tsc aperfmperf pni pclmulqdq dtes64 monitor ds_cpl vmx smx est tm2 ssse3 cx16 xtpr pdcm sse4_1 sse4_2 x2apic popcnt aes xsave avx lahf_lm ida arat epb xsaveopt pln pts dts tpr_shadow vnmi flexpriority ept vpid bogomips : 5587.63 clflush size : 64 cache_alignment : 64 address sizes : 36 bits physical, 48 bits virtual power management: I search here and found the answer to one post suggesting remove boot parameter 'nolapic'. However, on my particular laptop, ubuntu won't boot without this nolapic parameter. Is there anyway for ubuntu correly utility the full cpu power?

  Read the article

< Previous Page | 7 8 9 10 11 12 13 14 15 16 17 18  | Next Page >