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  • Query Performance Degrades with High Number of Logical Reads

    - by electricsk8
    I'm using Confio Ignite8 to derive this information, and monitor waits. I have one query that runs frequently, and I notice that on some days there is an extremely high number of logical reads incurred, +300,000,000 for 91,000 executions. On a good day, the logical reads are much lower, 18,000,000 for 94,000 executions. The execution plan for the query utilizes clustered index seeks, and is below. StmtText |--Nested Loops(Inner Join, OUTER REFERENCES:([f].[ParentId])) |--Clustered Index Seek(OBJECT:([StructuredFN].[dbo].[Folder].[PK_Folders] AS [f]), SEEK:([f].[FolderId]=(8125)), WHERE:([StructuredFN].[dbo].[Folder].[DealId] as [f].[DealId]=(300)) ORDERED FORWARD) |--Clustered Index Seek(OBJECT:([StructuredFN].[dbo].[Folder].[PK_Folders] AS [p]), SEEK:([p].[FolderId]=[StructuredFN].[dbo].[Folder].[ParentId] as [f].[ParentId]), WHERE:([StructuredFN].[dbo].[Folder].[DealId] as [p].[DealId]=(300)) ORDERED FORWARD) Output from showstatistics io ... Table 'Folder'. Scan count 0, logical reads 4, physical reads 0, read-ahead reads 0, lob logical reads 0, lob physical reads 0, lob read-ahead reads 0. Any ideas on how to troubleshoot where these high logical reads come from on certain days, and others nothing?

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  • Good resource for studying Database High Availability techniques

    - by Invincible
    Hello Can anybody suggest some good resource/book on Database high availability techniques? Moreover, High-availability of system software like Intrusion Prevention system or Web servers. I am considering high-availability is global term which covers clustring, cloud computing, replication, replica management, distributed synchronization for cluster. Thanks in advance!

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  • Synchronizing audio and video using MP4Box / ffmpeg to concatenate files

    - by jdl2003
    I have two H.264 encoded MPEG-4 files that I need to concatenate. I have been using MP4Box for this task by first ensuring the files are encoded identically (even went so far as to compare output from h264_parse on their video tracks) and then concatenating with this command: MP4Box -cat file1.mp4 -cat file2.mp4 output_file.mp4 This works and the output file is playable, but on playback in Quicktime or VLC the second video's audio starts too soon, making the entire second part of the concatenated file out of sync. I have tried reencoding the output through ffmpeg with -vcodec copy and -acodec copy but the sync issue persists. I have also tried converting to MPEG-2 format first, concatenating with a simple cat file1.mpg file2.mpg > output.mpg and reencoding the result with ffmpeg. This was even worse. I know that I can pass commands to MP4Box to adjust the start time of the audio track, but I am trying to do this programmatically for any input video (in the same encoding of course). I am looking for possible solutions that would not require human intervention / manual adjustments. Or, at least, an understanding of what is happening to make the second part of the concatenated video go out of sync.

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  • Sending microphone input over Remote Desktop 7.0

    - by Taylor Price
    I am using Remote Desktop 7 (the new version that came out with Windows 7) to control a Windows XP Pro machine. I have selected "Record from this computer" in the Remote Audio settings. When I connect to the machine, go to the control panel, open the sound panel, and go to the audio tab, I find that the default sound playback device is "Microsoft RDP Audio Driver". However, there is no default sound recording device. As expected, my IP phone thinks there is no recording device. If I am sitting in front of the computer with a mic plugged in, it works just fine. Has anybody else been able to get this work appropriately? Is there anything that I have to setup on the XP machine to get this working? Thanks in advance. Edit: As John T pointed out below, you have to be connecting to a Windows 7 Enterprise or Ultimate machine for this to work. I've also found out that Multi-monitor support has the same requirement.

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  • QuickTime Player sounds much better than iTunes

    - by Gene Goykhman
    I am playing a 320 kpbs encoded music MP3 in iTunes and the sound is substantially worse than the exact same file played back in QuickTime Player (Max OS X 10.8.5). I have maxed out system volume and iTunes playback volume. I have disabled all the audio processing features in iTunes (equalization, sound enhancer, etc.) The audio coming from iTunes still sounds resampled and/or processed, whereas QuickTime Player appears to be playing it "as is". Even when I Get Info on the MP3 file in Finder and play it back directly from the Get Info window it sounds good. It's just iTunes that seems to be mangling the song. I can notice a difference on virtually all my music, so it's not just one particular MP3. I suspect the issue is that iTunes is doing some kind of audio processing but I can't find a way to turn it off. This is the newest iTunes (11.1), but the problem has probably been going on for a while... I just switched to decent earbuds and started noticing the difference. What's the best way to force iTunes to play back the file as-is, or as close as possible to how QuickTime Player/Finder would play it?

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  • Speakers silent, headphones work in Ubuntu 9.04

    - by CarlF
    I'm running Ubuntu 9.04. Worked fine for months, then I rebooted yesterday after weeks of continuous operation. Now audio won't play through the speakers. The USB headset works fine, but the Conexant audio (CX20549) does not. Weirdly, it thinks it's playing. pavumeter shows appropriate levels, volume looks OK in alsamixer, but no sound. I did find this page: http://www.eugeneteplitsky.com/fixing-silent-pulseaudio-in-ubuntu-9-04/ Unfortunately the advice there doesn't help me. For one thing, the syntax for the alsa-base.conf file is apparently not actually documented anywhere. For another, my chipset isn't listed in the kernel.org docs he links to! EDIT: would upgrading to 9.10 help? Is there a major change in the audio subsystem between 9.04 and 9.10? Any suggestions? EDIT 2: This is stranger than I thought. Sound works normally in Xine, but is silent in Audacity, VLC and mplayer. What the?

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  • avconv gets killed if mkv has subtitles

    - by Lukas Knuth
    What I'm trying to do is to take a movie (in an Matroska container), convert all audio tracks to AC3 and don't touch anything else. I'm using this line: avconv -i infile.mkv -map 0 -vcodec copy -scodec copy -acodec ac3 -ab 256k outfile.mkv This works fine, except when there are subtitles embedded. Then, after some time processing with no progress, avconv just "dies" (output shortened, these seem to be the interesting parts): [matroska,webm @ 0xf867a0] max_analyze_duration reached [matroska,webm @ 0xf867a0] Estimating duration from bitrate, this may be inaccurate ... Incompatible sample format 's16' for codec 'ac3', auto-selecting format 'flt' ... Stream #0.0(eng): Video: H264 / 0x34363248, yuv420p, 1280x536 [PAR 1:1 DAR 160:67], q=2-31, 1k tbn, 1k tbc (default) Stream #0.1(ger): Audio: ac3, 48000 Hz, 5.1, flt, 256 kb/s (default) Stream #0.2(eng): Audio: ac3, 48000 Hz, 5.1, flt, 256 kb/s Stream #0.3(ger): Subtitle: dvdsub (default) (forced) Metadata: title : forced Stream #0.4(ger): Subtitle: dvdsub Metadata: title : complete Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (dca -> ac3) Stream #0:2 -> #0:2 (dca -> ac3) Stream #0:3 -> #0:3 (copy) Stream #0:4 -> #0:4 (copy) Input stream #0:2 frame changed from rate:48000 fmt:s16 ch:6 to rate:48000 fmt:flt ch:6 Input stream #0:1 frame changed from rate:48000 fmt:s16 ch:6 to rate:48000 fmt:flt ch:6 frame= 2606 fps=1303 q=-1.0 size= 3kB time=107.36 bitrate= 0.2kbits/s ... frame=96141 fps=813 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s frame=96251 fps=810 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s ... frame=97015 fps=397 q=-1.0 size= 2195806kB time=2807.04 bitrate=6408.2kbits/s Getötet ["Killed", in English] I have no idea why this happens, as there is no error-output. I'd like to just copy the subtitles over, not touch them at all. If that won't work, they can be completely dropped.

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  • Is ffmpeg incorrectly interpreting .aif files?

    - by marue
    Being on an Ubuntu 10.04 server i installed the ffmpeg packages with apt. ffmpeg is working afterwards, and doing as it should. Almost. For testing purposes i uploaded a few audiofiles. One of them, an aif file, is not being correctly interpreted. While on my workhorse (Mac SnowLeopard) ffmpeg tells the format as Stream #0.0: Audio: pcm_s24be, 44100 Hz, 2 channels, s32, 2116 kb/s my Ubuntu server says it is: Stream #0.0: Audio: pcm_s24be, 44100 Hz, stereo, s16, 2116 kb/s which is the wrong bitdepth. Ubuntu then fails to convert the file with the error message [pcm_s24be @ 0xcd4b580]invalid PCM packet Error while decoding stream #0.0 which certainly is not true. The file is perfectly valid. Are there any know issues for ffmpeg interpreting the aif format? How can i find out which version of the aif-codec ffmpeg is using? Any ideas how to approach this issue? ffprobe output: FFprobe version SVN-r20090707, Copyright (c) 2007-2009 Stefano Sabatini libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 built on Jan 20 2010 00:13:01, gcc: 4.4.3 20100116 (prerelease) Input #0, aiff, from 'testfile.aif': Duration: 00:00:04.00, start: 0.000000, bitrate: 2117 kb/s Stream #0.0: Audio: pcm_s24be, 44100 Hz, stereo, s16, 2116 kb/s update 2: Forcing the conversion with -sample_fmt s32 doesn't change anything. Strange thing is: Even without using -sample_fmt s32 i just realized that the conversion is working and creates valid audiofiles. There just is the error message from above.

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  • Problems with 5.1 digital out on Ubuntu 12.04

    - by user895319
    I've recently bought a new PC, installed Ubuntu and am now unable to get 5.1 digital sound working. Simple analogue stereo works fine on both the front and rear connectors. On my old box I connected the coax connection from my soundcard to my surround sound amplifier, set Settings-Sound to "Digital Stereo Duplex" and it worked. My old soundcard doesn't fit in my new machine so I'm using the built-in sound hardware. I'm connecting the combination output socket on the back of the PC via the same cable to my surround amp as before. The MB is an MSI Global H61M-P31 with an RealTek ALC887 sound chip. When I go to Settings-Sound I only see "Headphone Built-in Audio" and "Analogue Output Built-in Audio" - no digitial options. The output from aplay -l is: default Playback/recording through the PulseAudio sound server sysdefault:CARD=PCH HDA Intel PCH, ALC887-VD Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Front speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers dmix:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Direct sample mixing device dsnoop:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Direct sample snooping device hw:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Direct hardware device without any conversions plughw:CARD=PCH,DEV=0 HDA Intel PCH, ALC887-VD Analog Hardware device with all software conversions While googling for ALC887 I've seen some references to "ALC887 -VD Analog" and some to "ALC887 -VD Digital". Does anyone know if I need to force it to chance mode somehow? It's worth mentioning that when I set the output to 5.1 digital surround in Windows 7 on the same machine I still don't get any sound so it's not a unique Linux problem. Thanks for any help.

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  • 5.1 Surround Channels are Jumbled

    - by stickynips
    I had this exact setup working previously, but after a reformat it went screwy on me. I have an Onkyo A/V Receiver hooked up to my PC, via optical S/PDIF. Attached to the receiver is a 5.1 speaker setup (tested and working fine with my Xbox via the receiver). It seems to me that the audio channels are getting mixed up somehow between the PC and the receiver. I have a 5.1 test file which plays a sounds through each speaker individually. The channels are mixed as such: "Left Front" plays through my Right Front speaker "Center" plays through my Left Front speaker "Right Front" plays through my Center speaker "Left Rear" plays through my Subwoofer "Right Rear" plays through my Left Rear speaker I've tried downloading the latest Realtek HD Audio Drivers and the Realtek HD Audio Manager, but neither makes any difference. If there's a way I can manually rearrange the channels I believe it would fix the problem, but as far as I know this is impossible. edit: Sorry, I've forgotten some basic info. I'm running Windows 7 x64. The sound card is Realtek ALC892 embedded in a GIGABYTE GA-890GPA-UD3H AM3 motherboard.

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  • Tulsa SharePoint Interest Group – SharePoint 2010 Mini-Launch Event - Review

    - by dmccollough
    The Tulsa SharePoint Interest Group set a record for attendance last night at our SharePoint 2010 Mini-Launch Event. Approximately 40+ people showed up to listen to SharePoint MVP Eric Shupps, The SharePoint Cowboy to discuss all of the new features for both administrators and developers. All of the Tulsa SharePoint Interest Group Officers worked very hard to ensure that this event happened. We hosted our event at our local Dave & Busters and it was a great location with good food and great service. All of the officers of the Tulsa SharePoint Interest Group would like to extend a big Thank You to all of our sponsor that helped us in making our SharePoint 2010 Mini-Launch Event a reality.

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  • Which cloud hosting should I use? [closed]

    - by Alyssa Marie Isk
    Possible Duplicate: How to find web hosting that meets my requirements? If anyone wants to get some real life karma by giving a tiny non-profit pointers, please advise! I posted a thread about our website with highly variable traffic (www.WorldOceansDay.org). The event is on June 8th, and the traffic goes from 100-400/day in the off-season, to about 200,000 trying to access the site at any one time on June 8th. It's a Wordpress site hosted on GoDaddy shared hosting and predictably crashed horribly. From the internet's feedback, we've decided to move to a cloud server to handle the traffic, but I'm a huge newbie and I don't have very reliable mentorship, so I'm turning to crowdsourcing. We're trying to decide between Amazon Web Services and RackSpace Cloud servers. Our sys admin consultant also suggested GoDaddy's new 4GH but I have had such incredibly bad experiences with GoDaddy thus far that I am hesitant. From what I've read on the internet, RackSpace might be cheaper? Would AWS totally break the bank? We don't have a ton of money to spend on hosting. We'll also be using CloudFlare to cache and serve the pages since they're dynamic. I've found a few AWS & RackSpace calculators but I am not 100% on how to find those numbers... GoDaddy? Google Analytics? AWS calc is here: http://calculator.s3.amazonaws.com/calc5.html Rackspace is on the right: http://www.rackspace.com/cloud/cloud_hosting_products/servers/pricing/?0a313380 If anyone can help, or through some miracle feels like walking me through this, I would be incredibly appreciative.

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  • Our Oracle Recruitment Team is Growing - Multiple Job Opportunities in Bangalore, India

    - by david.talamelli
    DON"T GET STUCK IN THE MATRIXSEE YOUR FUTUREVISIT THE ORACLE The position(s): CORPORATE RECRUITING RESEARCH ANALYST(S) ABOUT ORACLE Oracle's business is information--how to manage it, use it, share it, protect it. For three decades, Oracle, the world's largest enterprise software company, has provided the software and services that allow organizations to get the most up-to-date and accurate information from their business systems. Only Oracle powers the information-driven enterprise by offering a complete, integrated solution for every segment of the process industry. When you run Oracle applications on Oracle technology, you speed implementation, optimize performance, and maximize ROI. Great hiring doesn't happen by accident; it's the culmination of a series of thoughtfully planned and well executed events. At the core of any hiring process is a sourcing strategy. This is where you come in... Do you want to be a part of a world-class recruiting organization that's on the cutting edge of technology? Would you like to experience a rewarding work environment that allows you to further develop your skills, while giving you the opportunity to develop new skills? If you answered yes, you've taken your first step towards a future with Oracle. We are building a Research Team to support our North America Recruitment Team, and we need creative, smart, and ambitious individuals to help us drive our research department forward. Oracle has a track record for employing and developing the very best in the industry. We invest generously in employee development, training and resources. Be a part of the most progressive internal recruiting team in the industry. For more information about Oracle, please visit our Web site at http://www.oracle.com Escape the hum drum job world matrix, visit the Oracle and be a part of a winning team, apply today. POSITION: Corporate Recruiting Research Analyst LOCATION: Bangalore, India RESPONSIBILITIES: •Develop candidate pipeline using Web 2.0 sourcing strategies and advanced Boolean Search techniques to support U.S. Recruiting Team for various job functions and levels. •Engage with assigned recruiters to understand the supported business as well as the recruiting requirements; partner with recruiters to meet expectations and deliver a qualified pipeline of candidates. •Source candidates to include both active and passive job seekers to provide a strong pipeline of qualified candidates for each recruiter; exercise creativity to find candidates using Oracle's advanced sourcing tools/techniques. •Fully evaluate candidate's background against the requirements provided by recruiter, and process leads using ATS (Applicant Tracking System). •Manage your efforts efficiently; maintain the highest levels of client satisfaction as well as strong operations and reporting of research activities. PREFERRED QUALIFICATIONS: •Fluent in English, with excellent written and oral communication skills. •Undergraduate degree required, MBA or Masters preferred. •Proficiency with Boolean Search techniques desired. •Ability to learn new software applications quickly. •Must be able to accommodate some U.S. evening hours. •Strong organization and attention to detail skills. •Prior HR or corporate in-house recruiting experiences a plus. •The fire in the belly to learn new ideas and succeed. •Ability to work in team and individual environments. This is an excellent opportunity to join Oracle in our Bangalore Offices. Interested applicants can send their resume to [email protected] or contact David on +61 3 8616 3364

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  • How to get the mic on the Creative X-Mod soundcard working correctly?

    - by Nyamiou The Galeanthrope
    Well, I have this problem for a while now. When my computer start the mic seem to work but it's like it's muted. I have to go to a terminal and type alsamixer -c 1 and then I set up PCM Capture Source on Line and set up it back to Mic to get the mic actually working. Is there is a way to do this automatically or to solve the problem. I use a special workaround on this card because of the bug #429642. My workaround is having this at the end of my /usr/share/pulseaudio/alsa-mixer/profile-sets/default.conf : [Mapping xmod-stereo-out] device-strings = surround51:%f description = Analog Stereo Creative Xmod channel-map = front-left,front-right paths-output = analog-output analog-output-headphones analog-output-mono analog-output-lfe-on-mono paths-input = analog-input analog-input-mic analog-input-linein analog-input-aux analog-input-video analog-input-tvtuner analog-input-fm analog-input-mic-line priority = 10 Maybe the bug come from here, maybe I have to change something. Thank you for any help.

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  • New Year's Resolution: Highest Availability at the Lowest Cost

    - by margaret.hamburger(at)oracle.com
    Don't miss this Webcast: Achieve 24/7 Cloud Availability Without Expensive Redundancy Event Date: 01/11/2011 10:00 AM Pacific Standard Time You'll learn how Oracle's Maximum Availability Architecture and Oracle Database 11g help you: Achieve the highest availability at the lowest cost Protect your systems from unplanned downtime Eliminate idle redundancy Register Now! var gaJsHost = (("https:" == document.location.protocol) ? "https://ssl." : "http://www."); document.write(unescape("%3Cscript src='" + gaJsHost + "google-analytics.com/ga.js' type='text/javascript'%3E%3C/script%3E")); try { var pageTracker = _gat._getTracker("UA-13185312-1"); pageTracker._trackPageview(); } catch(err) {}

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  • Renoise and dssi and jack

    - by Bojan
    This may be a little complicated, but, is jack a necessity? I mean, i use renoise, and, since i dont have the need for low latencies, do i really need to use it? My basic setup ( or workflow ) is that i use csound to render stuff to wav, then import it as a sample in renoise. That goes with field recordings, my own samples, etc. So, i dont need ultra low latencies, and i dont need to patch "cords", but i want to use dssi plugins, and dssi-vst. What would be something of a minimum requirements of apps that should work. Can renoise load dssi-vst plugins by itself or do i need to use jack to patch thru or something third, i tried to read lot of articles but i got lost in the different setups...

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  • Review the New Migration Guide to SQL Server 2012 Always On

    - by KKline
    I had the pleasure of meeting Mr. Cephas Lin, of Microsoft, last year at the SQL Saturday in Indianapolis and then later at the PASS Summit in the fall. Cephas has been writing content for SQL Server 2012 Always On. Cephas has recently published his first whitepaper, a migration guide to SQL Server AlwaysOn. Read it and then pass along any feedback: HERE Enjoy, -Kev - Follow me on Twitter !...(read more)

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  • Beat detection, weird detection

    - by Quincy
    I made this soundanalyzer class to detect beats in songs : // put it on pastebin for the big size, will put it here if people rather want that. pastebin.com/8PdgZPP3 but for some reason its only detecting beats from 637 sec to around 641(sec) and I have no idea why. I know the beats are being inserted from multiple bands since I am finding duplicates and it seems as its assigning a beat to each instant energy value in between those values. Its modeled after this : http://www.flipcode.com/misc/BeatDetectionAlgorithms.pdf So why won't the beats properly register ?

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  • Computer SOMETIMES recognizes when headphones are plugged in.

    - by rcrobot
    Whenever I plug my headphones into my computer's front headphone jack, I get a weird situation. Sometimes, the computer will recognize the headphones and work properly. But other times, the computer will play sound through both the headphones and my monitor's speaker. When this happens, the sound section of the system settings does not list the headphones. I can fix the issue temporarily by wiggling the headphone port, but if it gets wiggled the wrong way again, then the issue returns. My PC's case is a Rosewill Challenger. I have tried multiple headphones and the same issue is there. I suspect that this might be a hardware related issue, but if there is any way to fix it with software, that would be helpful. This is what it looks like when everything is working properly: This happens when I wiggle the headphone port. I can quickly switch between these two by doing so:

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  • How to correctly Dispose a SourceVoice once its finished

    - by clamp
    i am starting to play a sound with XAudio2 and SourceVoice and once its finished, it should be correctly disposed to not have any leaks. i was expecting it to be something like this: sourceVoice.Start(); sourceVoice.StreamEnd += delegate { if (!sourceVoice.IsDisposed) { sourceVoice.DestroyVoice(); sourceVoice.Dispose(); } }; but that crashes with a read access violation in native code deep in XAudio2.dll which i cant debug.

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  • Playing a Song causing WP7 to crash on phone, but not on emulator

    - by Michael Zehnich
    Hi there, I am trying to implement a song into a game that begins playing and continually loops on Windows Phone 7 via XNA 4.0. On the emulator, this works fine, however when deployed to a phone, it simply gives a black screen before going back to the home screen. Here is the rogue code in question, and commenting this code out makes the app run fine on the phone: // in the constructor fields private Song song; // in the LoadContent() method song = Content.Load<Song>("song"); // in the Update() method if (MediaPlayer.GameHasControl && MediaPlayer.State != MediaState.Playing) { MediaPlayer.Play(song); } The song file itself is a 2:53 long, 2.28mb .wma file at 106kbps bitrate. Again this works perfectly on emulator but does not run at all on phone. Thanks for any help you can provide!

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  • Is the SAN dying???

    - by RickHeiges
    Is the SAN dying? The reason that I ask this question is that MSFT has unleashed technologies this year that point in that direction Always ON Availability Groups shuns shared storage Windows 2012 has Storage Replication Technology that does not require a SAN Windows 2012 has Hyper-V Replica Technology that does not require a SAN PDW v2 continues to reinforce the approach to avoid shared storage I'm not saying that SAN technology does not have its place or does not have benefits inherent to the beast....(read more)

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  • How much is modern programming still tied to underyling digital logic?

    - by New Talk
    First of all: I've got no academic background. I'm working primarily with Java and Spring and I'm also fond of web programming and relational databases. I hope I'm using the right terms and I hope that this vague question makes some sense. Today the following question came to my mind: How much is modern programming still tied to the underlying digital logic? With modern programming I mean concepts like OOP, AOP, Java 7, AJAX, … I hope you get the idea. Do they no longer need the digital logic with which computers are working internally? Or is binary logic still ubiquitous when programming this way? If I'd change the inner workings of a computer overnight, would it matter, because my programming techniques are already that abstract? P. S.: With digital logic I mean the physical representation of everything "inside" the computer as zeroes and ones. Changed "binary" to "digital".

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  • How do I get my ART USB Dual Pre preamp to work?

    - by Zach
    I am using Audacity. I have an ART USB Dual Pre preamp. Ubuntu is not recognizing it whatsoever. I am able to record in Audacity, but it is using the mic that is built into my computer (which is a compaq Presario CQ50) instead of the one plugged into the preamp. How do I get Ubuntu to recognize the preamp that is plugged into my computer? Something tells me it has to do with the installation of the preamp software. It came with a installation CD, but when I go to "install", the nothing happens. I can view what is on the CD, but there is no installing of anything. Please help!

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  • Pain of the Week/Expert's Perspective: Performance Tuning for Backups and Restores

    - by KKline
    First off - the Pain of the Week webcast series has been renamed. It's now known as The Expert's Perspective . Please join us for future webcasts and, if you're interested in speaking, drop me a note to see if we can get you on the roster! The bigger your databases get, the longer backups take. That doesn't really seem like a huge problem — until disaster strikes and you need to restore your databases as fast as possible. Join my buddy Brent Ozar ( blog | twitter ), a Microsoft Certified Master of...(read more)

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