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  • 1. Rename computer 2. Add this computer

    - by user91583
    I've followed the instructions but they haven't worked. How do I -1. Rename this computer as it would appear in: -a Ubuntu One Preferences -- Devices; and [next line] -b https://one.ubuntu.com/account/ -- Devices -2. Your https://one.ubuntu.com/help/tutorial/install-and-setup-ubuntu-one-8/ says After you sign in to Ubuntu One via your web browser you should see the Add this Computer button. I am on the unfortunate side of that 'should'. There is no Add this Computer button.

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  • Get specifc value from JSon string using JSon.Net

    - by dean nolan
    I am trying to get a value from a JSon formatted string. It was to get album info from a website called Freebase. My result is like this: { "a0": { "code": "/api/status/error", "messages": [ { "code": "/api/status/error/mql/result", "info": { "count": 20, "result": [ { "album": [ { "name": "Definitely Maybe", "release_date": "1994-08-30" } ], "artist": "Oasis", "name": "Live Forever", "type": "/music/track" }, { "album": [ { "name": "Most Wanted Rock 1", "release_date": null } ], "artist": "Oasis", "name": "Live Forever", "type": "/music/track" }, { "album": [ { "name": "Alternative 90s", "release_date": null } ], "artist": "Oasis", "name": "Live Forever", "type": "/music/track" }, { "album": [ { "name": "Live Forever: Best of Britpop", "release_date": "2003-03-03" } ], "artist": "Oasis", "name": "Live Forever", "type": "/music/track" }, { "album": [ { "name": "The Best... In the World... Ever! Volume 5", "release_date": "1997-03-31" } ], "artist": "Oasis", "name": "Live Forever", "type": "/music/track" }, { "album": [ { "name": "Live 4 Ever", "release_date": "1998-06-29" } ], "artist": "Oasis", "name": "Live Forever", "type": "/music/track" }, { "album": [ { "name": "De Afrekening, Volume 8", "release_date": "1994" } ], "artist": "Oasis", "name": "Live Forever", "type": "/music/track" }, { "album": [ { "name": "Now That's What I Call Music! 33", "release_date": null } ], "artist": "Oasis", "name": "Live Forever", "type": "/music/track" }, { "album": [ { "name": "Q: Anthems", "release_date": null } ], "artist": "Oasis", "name": "Live Forever", "type": "/music/track" }, { "album": [ { "name": "The Best Anthems... Ever! Volume 2", "release_date": null } ], "artist": "Oasis", "name": "Live Forever", "type": "/music/track" }, { "album": [ { "name": "1995 Mercury Music Prize: Ten Albums of the Year", "release_date": null } ], "artist": "Oasis", "name": "Live Forever", "type": "/music/track" }, { "album": [ { "name": "Now That's What I Call Music! 1994", "release_date": null } ], "artist": "Oasis", "name": "Live Forever", "type": "/music/track" }, { "album": [ { "name": "Indie Top 20, Volume 21", "release_date": "1995" } ], "artist": "Oasis", "name": "Live Forever", "type": "/music/track" }, { "album": [ { "name": "Dad Rocks!", "release_date": "2006-06-05" } ], "artist": "Oasis", "name": "Live Forever", "type": "/music/track" }, { "album": [ { "name": "Untitled", "release_date": null } ], "artist": "Oasis", "name": "Live Forever", "type": "/music/track" }, { "album": [ { "name": "The Greatest Hits of 1994", "release_date": "1994-10" } ], "artist": "Oasis", "name": "Live Forever", "type": "/music/track" }, { "album": [ { "name": "Top of the Pops 2", "release_date": "2000-03-27" } ], "artist": "Oasis", "name": "Live Forever", "type": "/music/track" }, { "album": [ { "name": "Q: Anthems (disc 1)", "release_date": null } ], "artist": "Oasis", "name": "Live Forever", "type": "/music/track" }, { "album": [ { "name": "Jamie Oliver's Cookin'", "release_date": "2001" } ], "artist": "Oasis", "name": "Live Forever", "type": "/music/track" }, { "album": [ { "name": "Killer Buzz", "release_date": "2001" } ], "artist": "Oasis", "name": "Live Forever", "type": "/music/track" } ] }, "message": "Unique query may have at most one result. Got 20", "path": "", "query": { "album": [ { "name": null, "release_date": null, "sort": "release_date" } ], "artist": "Oasis", "error_inside": ".", "name": "Live Forever", "type": "/music/track" } } ] }, "code": "/api/status/ok", "status": "200 OK", "transaction_id": "cache;cache04.p01.sjc1:8101;2010-03-30T18:04:20Z;0035" } I am looking to get the first album title, Definitely Maybe, from this list. I have tried parsing the string like this: JObject o = JObject.Parse(jsonString); string album = (string)o[""]; But no matter what I have tried I don't know what to put in those quotes. How would I get this specific value or be able to search for it? Thanks

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  • filesystem mounting problem

    - by user306988
    Hello, A DAS box is attached to my linux box using LSI SCSI HBA. Volume is properly detected on the linux box and filesystem is created using mkfs.ext2 /dev/sdc #No partition table I can not mount the volume using mount/dev/sdc /mnt/temp -t ext3 But I can mount it using mount /dev/sdc /mnt/temp -t ext3 -o loop Can anybody please tell me what "-o loop" option does internally? Has anybody faced this option before? Thanks in advance, prashant

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  • How to add my program to the OS X system menu bar?

    - by Joe
    I have created a volume controller for iTunes but I would like this app to place an icon on the OS X system menu bar and have my slider controller drop down. I created this because I have to switch to iTunes to change the volume of the music because I am using the digital-out audio and the keyboard keys do not work in digital-out mode. Any guidance would be greatly appreciated.

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  • Silverlight audio components

    - by Shurup
    Do you know any good commertial or free audio / sound components that can be used in Silverlight (better on the client)? Desired features: visualizing (waveform, peakform etc., zoom, editing volume, pan, start and end markers etc.) editing (channels: volume, pan, mute, fading, etc., merge, cut, copy, paste, etc., effects) mixing different sound tracks, creating a downmix exporting result in some in Silverlight usable (compressed) formats Some useful projects that I've founded: Saluse MediaKit for Silverlight (MP3 decoder, AudioPreProcessor, visualizers, effects) Microsoft Design Toolbox sound mixer (incl. source code and video) and Demo

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  • Custom preference screen

    - by rantravee
    Hi, I have a slider for the user to control the volume of my application. What I'd like to have is a icon or something else similar to "volume display" like in the next picture at the slider summary. Is this possible ? How could this be accomplished ?

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  • How to adjust microphone gain from C# (needs to work on XP & W7)...

    - by Ed
    First, note that I know there are a few questions like this already posted; however they don't seem to address the problem adequately. I have a C# application, with all the pInvoke hooks to talk to the waveXXX API, and I'm able to do capture and play back of audio with that. I'm also able to adjust speaker (WaveOut) volume with that API. The problem is that for whatever reason, that API does not allow me to adjust microphone (WaveIn) volume. So, I managed to find some mixer code that I've also pulled in and access through pInvoke and that allows me to adjust microphone volume, but only on my W7 PC. The mixer code I started with comes from here: http://social.msdn.microsoft.com/Forums/en-US/isvvba/thread/05dc2d35-1d45-4837-8e16-562ee919da85 and it works, but is written to adjust speaker volume. I added the SetMicVolume method shown here... public static void SetMicVolume(int mxid, int percentage) { bool rc; int mixer, vVolume; MIXERCONTROL volCtrl = new MIXERCONTROL(); int currentVol; mixerOpen(out mixer, mxid, 0, 0, MIXER_OBJECTF_WAVEIN); int type = MIXERCONTROL_CONTROLTYPE_VOLUME; rc = GetVolumeControl(mixer, MIXERLINE_COMPONENTTYPE_SRC_MICROPHONE, type, out volCtrl, out currentVol); if (rc == false) { mixerClose(mixer); mixerOpen(out mixer, 0, 0, 0, 0); rc = GetVolumeControl(mixer, MIXERLINE_COMPONENTTYPE_SRC_MICROPHONE, type, out volCtrl, out currentVol); if (rc == false) throw new Exception("SetMicVolume/GetVolumeControl() failed"); } vVolume = ((int)((float)(volCtrl.lMaximum - volCtrl.lMinimum) / 100.0F) * percentage); rc = SetVolumeControl(mixer, volCtrl, vVolume); if (rc == false) throw new Exception("SetMicVolume/SetVolumeControl() failed"); mixerClose(mixer); } Note the "second attempt" to call GetVolumeControl(). This is done because on XP, in the first call to GetVolumeControl (refer to site above for that code), the call to mixerGetLineControlsA() fails with XP systems returning MIXERR_INVALCONTROL. Then, with this second attempt using mixerOpen(out mixer, 0, 0, 0, 0), the code doesn't return a failure but the mic gain is unaffected. Note, as I said above, this works on W7 (the second attempt is never executed because it doesn't fail using mixerOpen(out mixer, mxid, 0, 0, MIXER_OBJECTF_WAVEIN)). I admit to not having a good grasp on the mixer API, so that's what I'm looking into now; however if anyone has a clue why this would work on W7, but not XP, I'd sure like to hear it. Meanwhile, if I figure it out before I get a response, I'll post my own answer...

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  • Why is this simple Mobile Form not closed when using the player

    - by ajhvdb
    Hi, I created this simple sample Form with the close button. Everything is working as expected when NOT using the Interop.WMPLib.dll I've seen other applications using this without problems but why isn't the Form process closed when I just add the line: SoundPlayer myPlayer = new SoundPlayer(); and of course dispose it: if (myPlayer != null) { myPlayer.Dispose(); myPlayer = null; } The Form closes but the debugger VS2008 is still active. The Form project and the dll are still active. If you send me an email to [email protected], I can send you the zipped project. Below is the class for the dll: using System; using System.Collections.Generic; using System.Text; using System.Threading; using System.Runtime.InteropServices; using WMPLib; namespace WindowsMobile.Utilities { public delegate void SoundPlayerStateChanged(SoundPlayer sender, SoundPlayerState newState); public enum SoundPlayerState { Stopped, Playing, Paused, } public class SoundPlayer : IDisposable { [DllImport("coredll")] public extern static int waveOutSetVolume(int hwo, uint dwVolume); [DllImport("coredll")] public extern static int waveOutGetVolume(int hwo, out uint dwVolume); WindowsMediaPlayer myPlayer = new WindowsMediaPlayer(); public SoundPlayer() { myPlayer.uiMode = "invisible"; myPlayer.settings.volume = 100; } string mySoundLocation = string.Empty; public string SoundLocation { get { return mySoundLocation; } set { mySoundLocation = value; } } public void Pause() { myPlayer.controls.pause(); } public void PlayLooping() { Stop(); myPlayer.URL = mySoundLocation; myPlayer.settings.setMode("loop", true); } public int Volume { get { return myPlayer.settings.volume; } set { myPlayer.settings.volume = value; } } public void Play() { Stop(); myPlayer.URL = mySoundLocation; myPlayer.controls.play(); } public void Stop() { myPlayer.controls.stop(); myPlayer.close(); } #region IDisposable Members public void Dispose() { try { Stop(); } catch (Exception) { } // need this otherwise the process won't exit?! try { int ret = Marshal.FinalReleaseComObject(myPlayer); } catch (Exception) { } myPlayer = null; GC.Collect(); } #endregion } }

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  • command to format mount points in windows 2003

    - by user136104
    I need a command to format the mount point in windows 2003. Generally we use the following command to format a volume in windows type C:\sri.txt| FORMAT E: /v:volume /Q /fs:NTFS But by using the same command we can't format the mount points. So is there any command to format a mount point in windows

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  • Does anyone really understand how HFSC scheduling in Linux/BSD works?

    - by Mecki
    I read the original SIGCOMM '97 PostScript paper about HFSC, it is very technically, but I understand the basic concept. Instead of giving a linear service curve (as with pretty much every other scheduling algorithm), you can specify a convex or concave service curve and thus it is possible to decouple bandwidth and delay. However, even though this paper mentions to kind of scheduling algorithms being used (real-time and link-share), it always only mentions ONE curve per scheduling class (the decoupling is done by specifying this curve, only one curve is needed for that). Now HFSC has been implemented for BSD (OpenBSD, FreeBSD, etc.) using the ALTQ scheduling framework and it has been implemented Linux using the TC scheduling framework (part of iproute2). Both implementations added two additional service curves, that were NOT in the original paper! A real-time service curve and an upper-limit service curve. Again, please note that the original paper mentions two scheduling algorithms (real-time and link-share), but in that paper both work with one single service curve. There never have been two independent service curves for either one as you currently find in BSD and Linux. Even worse, some version of ALTQ seems to add an additional queue priority to HSFC (there is no such thing as priority in the original paper either). I found several BSD HowTo's mentioning this priority setting (even though the man page of the latest ALTQ release knows no such parameter for HSFC, so officially it does not even exist). This all makes the HFSC scheduling even more complex than the algorithm described in the original paper and there are tons of tutorials on the Internet that often contradict each other, one claiming the opposite of the other one. This is probably the main reason why nobody really seems to understand how HFSC scheduling really works. Before I can ask my questions, we need a sample setup of some kind. I'll use a very simple one as seen in the image below: Here are some questions I cannot answer because the tutorials contradict each other: What for do I need a real-time curve at all? Assuming A1, A2, B1, B2 are all 128 kbit/s link-share (no real-time curve for either one), then each of those will get 128 kbit/s if the root has 512 kbit/s to distribute (and A and B are both 256 kbit/s of course), right? Why would I additionally give A1 and B1 a real-time curve with 128 kbit/s? What would this be good for? To give those two a higher priority? According to original paper I can give them a higher priority by using a curve, that's what HFSC is all about after all. By giving both classes a curve of [256kbit/s 20ms 128kbit/s] both have twice the priority than A2 and B2 automatically (still only getting 128 kbit/s on average) Does the real-time bandwidth count towards the link-share bandwidth? E.g. if A1 and B1 both only have 64kbit/s real-time and 64kbit/s link-share bandwidth, does that mean once they are served 64kbit/s via real-time, their link-share requirement is satisfied as well (they might get excess bandwidth, but lets ignore that for a second) or does that mean they get another 64 kbit/s via link-share? So does each class has a bandwidth "requirement" of real-time plus link-share? Or does a class only have a higher requirement than the real-time curve if the link-share curve is higher than the real-time curve (current link-share requirement equals specified link-share requirement minus real-time bandwidth already provided to this class)? Is upper limit curve applied to real-time as well, only to link-share, or maybe to both? Some tutorials say one way, some say the other way. Some even claim upper-limit is the maximum for real-time bandwidth + link-share bandwidth? What is the truth? Assuming A2 and B2 are both 128 kbit/s, does it make any difference if A1 and B1 are 128 kbit/s link-share only, or 64 kbit/s real-time and 128 kbit/s link-share, and if so, what difference? If I use the seperate real-time curve to increase priorities of classes, why would I need "curves" at all? Why is not real-time a flat value and link-share also a flat value? Why are both curves? The need for curves is clear in the original paper, because there is only one attribute of that kind per class. But now, having three attributes (real-time, link-share, and upper-limit) what for do I still need curves on each one? Why would I want the curves shape (not average bandwidth, but their slopes) to be different for real-time and link-share traffic? According to the little documentation available, real-time curve values are totally ignored for inner classes (class A and B), they are only applied to leaf classes (A1, A2, B1, B2). If that is true, why does the ALTQ HFSC sample configuration (search for 3.3 Sample configuration) set real-time curves on inner classes and claims that those set the guaranteed rate of those inner classes? Isn't that completely pointless? (note: pshare sets the link-share curve in ALTQ and grate the real-time curve; you can see this in the paragraph above the sample configuration). Some tutorials say the sum of all real-time curves may not be higher than 80% of the line speed, others say it must not be higher than 70% of the line speed. Which one is right or are they maybe both wrong? One tutorial said you shall forget all the theory. No matter how things really work (schedulers and bandwidth distribution), imagine the three curves according to the following "simplified mind model": real-time is the guaranteed bandwidth that this class will always get. link-share is the bandwidth that this class wants to become fully satisfied, but satisfaction cannot be guaranteed. In case there is excess bandwidth, the class might even get offered more bandwidth than necessary to become satisfied, but it may never use more than upper-limit says. For all this to work, the sum of all real-time bandwidths may not be above xx% of the line speed (see question above, the percentage varies). Question: Is this more or less accurate or a total misunderstanding of HSFC? And if assumption above is really accurate, where is prioritization in that model? E.g. every class might have a real-time bandwidth (guaranteed), a link-share bandwidth (not guaranteed) and an maybe an upper-limit, but still some classes have higher priority needs than other classes. In that case I must still prioritize somehow, even among real-time traffic of those classes. Would I prioritize by the slope of the curves? And if so, which curve? The real-time curve? The link-share curve? The upper-limit curve? All of them? Would I give all of them the same slope or each a different one and how to find out the right slope? I still haven't lost hope that there exists at least a hand full of people in this world that really understood HFSC and are able to answer all these questions accurately. And doing so without contradicting each other in the answers would be really nice ;-)

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  • Does anyone really understand how HFSC scheduling in Linux/BSD works?

    - by Mecki
    I read the original SIGCOMM '97 PostScript paper about HFSC, it is very technically, but I understand the basic concept. Instead of giving a linear service curve (as with pretty much every other scheduling algorithm), you can specify a convex or concave service curve and thus it is possible to decouple bandwidth and delay. However, even though this paper mentions to kind of scheduling algorithms being used (real-time and link-share), it always only mentions ONE curve per scheduling class (the decoupling is done by specifying this curve, only one curve is needed for that). Now HFSC has been implemented for BSD (OpenBSD, FreeBSD, etc.) using the ALTQ scheduling framework and it has been implemented Linux using the TC scheduling framework (part of iproute2). Both implementations added two additional service curves, that were NOT in the original paper! A real-time service curve and an upper-limit service curve. Again, please note that the original paper mentions two scheduling algorithms (real-time and link-share), but in that paper both work with one single service curve. There never have been two independent service curves for either one as you currently find in BSD and Linux. Even worse, some version of ALTQ seems to add an additional queue priority to HSFC (there is no such thing as priority in the original paper either). I found several BSD HowTo's mentioning this priority setting (even though the man page of the latest ALTQ release knows no such parameter for HSFC, so officially it does not even exist). This all makes the HFSC scheduling even more complex than the algorithm described in the original paper and there are tons of tutorials on the Internet that often contradict each other, one claiming the opposite of the other one. This is probably the main reason why nobody really seems to understand how HFSC scheduling really works. Before I can ask my questions, we need a sample setup of some kind. I'll use a very simple one as seen in the image below: Here are some questions I cannot answer because the tutorials contradict each other: What for do I need a real-time curve at all? Assuming A1, A2, B1, B2 are all 128 kbit/s link-share (no real-time curve for either one), then each of those will get 128 kbit/s if the root has 512 kbit/s to distribute (and A and B are both 256 kbit/s of course), right? Why would I additionally give A1 and B1 a real-time curve with 128 kbit/s? What would this be good for? To give those two a higher priority? According to original paper I can give them a higher priority by using a curve, that's what HFSC is all about after all. By giving both classes a curve of [256kbit/s 20ms 128kbit/s] both have twice the priority than A2 and B2 automatically (still only getting 128 kbit/s on average) Does the real-time bandwidth count towards the link-share bandwidth? E.g. if A1 and B1 both only have 64kbit/s real-time and 64kbit/s link-share bandwidth, does that mean once they are served 64kbit/s via real-time, their link-share requirement is satisfied as well (they might get excess bandwidth, but lets ignore that for a second) or does that mean they get another 64 kbit/s via link-share? So does each class has a bandwidth "requirement" of real-time plus link-share? Or does a class only have a higher requirement than the real-time curve if the link-share curve is higher than the real-time curve (current link-share requirement equals specified link-share requirement minus real-time bandwidth already provided to this class)? Is upper limit curve applied to real-time as well, only to link-share, or maybe to both? Some tutorials say one way, some say the other way. Some even claim upper-limit is the maximum for real-time bandwidth + link-share bandwidth? What is the truth? Assuming A2 and B2 are both 128 kbit/s, does it make any difference if A1 and B1 are 128 kbit/s link-share only, or 64 kbit/s real-time and 128 kbit/s link-share, and if so, what difference? If I use the seperate real-time curve to increase priorities of classes, why would I need "curves" at all? Why is not real-time a flat value and link-share also a flat value? Why are both curves? The need for curves is clear in the original paper, because there is only one attribute of that kind per class. But now, having three attributes (real-time, link-share, and upper-limit) what for do I still need curves on each one? Why would I want the curves shape (not average bandwidth, but their slopes) to be different for real-time and link-share traffic? According to the little documentation available, real-time curve values are totally ignored for inner classes (class A and B), they are only applied to leaf classes (A1, A2, B1, B2). If that is true, why does the ALTQ HFSC sample configuration (search for 3.3 Sample configuration) set real-time curves on inner classes and claims that those set the guaranteed rate of those inner classes? Isn't that completely pointless? (note: pshare sets the link-share curve in ALTQ and grate the real-time curve; you can see this in the paragraph above the sample configuration). Some tutorials say the sum of all real-time curves may not be higher than 80% of the line speed, others say it must not be higher than 70% of the line speed. Which one is right or are they maybe both wrong? One tutorial said you shall forget all the theory. No matter how things really work (schedulers and bandwidth distribution), imagine the three curves according to the following "simplified mind model": real-time is the guaranteed bandwidth that this class will always get. link-share is the bandwidth that this class wants to become fully satisfied, but satisfaction cannot be guaranteed. In case there is excess bandwidth, the class might even get offered more bandwidth than necessary to become satisfied, but it may never use more than upper-limit says. For all this to work, the sum of all real-time bandwidths may not be above xx% of the line speed (see question above, the percentage varies). Question: Is this more or less accurate or a total misunderstanding of HSFC? And if assumption above is really accurate, where is prioritization in that model? E.g. every class might have a real-time bandwidth (guaranteed), a link-share bandwidth (not guaranteed) and an maybe an upper-limit, but still some classes have higher priority needs than other classes. In that case I must still prioritize somehow, even among real-time traffic of those classes. Would I prioritize by the slope of the curves? And if so, which curve? The real-time curve? The link-share curve? The upper-limit curve? All of them? Would I give all of them the same slope or each a different one and how to find out the right slope? I still haven't lost hope that there exists at least a hand full of people in this world that really understood HFSC and are able to answer all these questions accurately. And doing so without contradicting each other in the answers would be really nice ;-)

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  • what am I doing wrong? Trying to reinstall Windows 7 starter onto my Acer Aspire One Netbook?

    - by Robbie Roberts
    I have been having some issues with my Netbook so I figured I would reformat it. I downloaded a copy of windows 7 starter, inserted it into my usb dvd drive and started my netbook. I made it as far as, "where do you want to install windows?" and it seems like the computer just freezes. it shows, Disk 0 Partition 1: PQSERVICE 13.0 GB OEM (Reserved) Disk 0 Partition 2: SYSTEM RESERVED 101.0 MB System Disk 0 Partition 3 218.8 GB Primary I cannot click on either of them, What am I doing wrong?

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  • PC won't boot with 4 prong CPU connector connected. Motherboard or CPU dead - which one? Both?

    - by scrot
    In the middle of use, my computer shut down. I tried to turn it on but it maybe comes on for half a second, shuts down, then flickers again very quickly about 2 seconds later before giving up completely. So I took everything apart and nailed it down to the power supply, the mobo, or the CPU. I had an extra old power supply around, hooked that up and it had the same problems, so that can't be it. When the 4 prong CPU connector is not connected, the motherboard 'functions' in that it stays on and runs the heat sink etc. Throw in the 4 prong connector and that's when it doesn't stay on. So it must be the CPU, right? Motherboard: ASUS P6X58D CPU: Intel i7-920 Bot are under warranty (supposedly) - should I get both replaced or what?

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  • Running two Magentos installations, one of which has 3 stores set up as multi-store. Which server?

    - by Pedro Peixoto
    I want to run 4 Magento stores in 2 different installations. 1 is a standalonne installation with 3 languages. The other is a multi-store with 3 different online stores in different domains. At the moment we have a VPS with 1GB memory, would that be enough? I ask because I've finished the standalone store and already put it online, and the server is already running on 62% memory. The ideal would be that this is enough as my company wouldn't like to move to a Dedicated Server (as it involves costs). I'm sure I can try to optimize Magento to run on lower memory (I'm expecting visits averaging 2000/day on all sites), if I could have some tips on the best way to do that Id appreciate it too.

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  • Should the hostname of my VPS point to the dedi IP of my Domain or to to a shared one used for new account creation?

    - by thomas
    I leased a VPS which I want to use to sell shared hosting. 3 IPs - I call them A, B and C here for simplicity. Actual setup is: A=NS1.mydomain.com; host.mydomain.com and is used to set-up new accounts in shared environment B=NS2.mydomain.com C=dedicated IP for mydomain.com (SSL secured) The more I read about DNS, the more I get confused; thus my question: Is this configuration "Good Practice", especially the hostname pointing to A rather than to C? And what would be a better alternative?

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  • I overwrote a large file with a blank one on a linux server. Can I recover the existing file?

    - by user39234
    I came back to my machine, tried saving a file over ssh onto my linux server (CentOS). It failed. I wasn't interested in keeping any changes I had made so I closed my editor and reopened the file (over ssh). The save attempt wiped the file. I have made loads of changes to it since I last uploaded to revision control. Seeing as it has just wiped the file I assume the data is still there. It's just a text file (php), is there any way of recovering it?

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  • I'm having two Windows 7 installed and running in my laptop. How can I delete the other one?

    - by mandy
    I have another concern with my laptop. When I had a problem with the microphone, I tried to restore the system. But sadly, I had a mistake with the process and accidentally it was formatted. Its a built in laptop. (Toshiba m840, Windows 7, 64 bit) After the format process, programs were automatically installed. Until a friend of mine told me that I might have 2 Windows installed already. When we checked the program, he was right. I'm having 2 program files folders, which means I really have 2 Windows installed and running in my laptop. Supposed to be, there should be only 1 right? Can anyone help me how to delete the other Windows? Because I think it is occupying much space in the hard drive. They say it could make my laptop become slower.

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  • How to reduce iOS AVPlayer start delay

    - by Bernt Habermeier
    Note, for the below question: All assets are local on the device -- no network streaming is taking place. The videos contain audio tracks. I'm working on an iOS application that requires playing video files with minimum delay to start the video clip in question. Unfortunately we do not know what specific video clip is next until we actually need to start it up. Specifically: When one video clip is playing, we will know what the next set of (roughly) 10 video clips are, but we don't know which one exactly, until it comes time to 'immediately' play the next clip. What I've done to look at actual start delays is to call addBoundaryTimeObserverForTimes on the video player, with a time period of one millisecond to see when the video actually started to play, and I take the difference of that time stamp with the first place in the code that indicates which asset to start playing. From what I've seen thus-far, I have found that using the combination of AVAsset loading, and then creating an AVPlayerItem from that once it's ready, and then waiting for AVPlayerStatusReadyToPlay before I call play, tends to take between 1 and 3 seconds to start the clip. I've since switched to what I think is roughly equivalent: calling [AVPlayerItem playerItemWithURL:] and waiting for AVPlayerItemStatusReadyToPlay to play. Roughly same performance. One thing I'm observing is that the first AVPlayer item load is slower than the rest. Seems one idea is to pre-flight the AVPlayer with a short / empty asset before trying to play the first video might be of good general practice. [http://stackoverflow.com/questions/900461/slow-start-for-avaudioplayer-the-first-time-a-sound-is-played] I'd love to get the video start times down as much as possible, and have some ideas of things to experiment with, but would like some guidance from anyone that might be able to help. Update: idea 7, below, as-implemented yields switching times of around 500 ms. This is an improvement, but it it'd be nice to get this even faster. Idea 1: Use N AVPlayers (won't work) Using ~ 10 AVPPlayer objects and start-and-pause all ~ 10 clips, and once we know which one we really need, switch to, and un-pause the correct AVPlayer, and start all over again for the next cycle. I don't think this works, because I've read there is roughly a limit of 4 active AVPlayer's in iOS. There was someone asking about this on StackOverflow here, and found out about the 4 AVPlayer limit: fast-switching-between-videos-using-avfoundation Idea 2: Use AVQueuePlayer (won't work) I don't believe that shoving 10 AVPlayerItems into an AVQueuePlayer would pre-load them all for seamless start. AVQueuePlayer is a queue, and I think it really only makes the next video in the queue ready for immediate playback. I don't know which one out of ~10 videos we do want to play back, until it's time to start that one. ios-avplayer-video-preloading Idea 3: Load, Play, and retain AVPlayerItems in background (not 100% sure yet -- but not looking good) I'm looking at if there is any benefit to load and play the first second of each video clip in the background (suppress video and audio output), and keep a reference to each AVPlayerItem, and when we know which item needs to be played for real, swap that one in, and swap the background AVPlayer with the active one. Rinse and Repeat. The theory would be that recently played AVPlayer/AVPlayerItem's may still hold some prepared resources which would make subsequent playback faster. So far, I have not seen benefits from this, but I might not have the AVPlayerLayer setup correctly for the background. I doubt this will really improve things from what I've seen. Idea 4: Use a different file format -- maybe one that is faster to load? I'm currently using .m4v's (video-MPEG4) H.264 format. I have not played around with other formats, but it may well be that some formats are faster to decode / get ready than others. Possible still using video-MPEG4 but with a different codec, or maybe quicktime? Maybe a lossless video format where decoding / setup is faster? Idea 5: Combination of lossless video format + AVQueuePlayer If there is a video format that is fast to load, but maybe where the file size is insane, one idea might be to pre-prepare the first 10 seconds of each video clip with a version that is boated but faster to load, but back that up with an asset that is encoded in H.264. Use an AVQueuePlayer, and add the first 10 seconds in the uncompressed file format, and follow that up with one that is in H.264 which gets up to 10 seconds of prepare/preload time. So I'd get 'the best' of both worlds: fast start times, but also benefits from a more compact format. Idea 6: Use a non-standard AVPlayer / write my own / use someone else's Given my needs, maybe I can't use AVPlayer, but have to resort to AVAssetReader, and decode the first few seconds (possibly write raw file to disk), and when it comes to playback, make use of the raw format to play it back fast. Seems like a huge project to me, and if I go about it in a naive way, it's unclear / unlikely to even work better. Each decoded and uncompressed video frame is 2.25 MB. Naively speaking -- if we go with ~ 30 fps for the video, I'd end up with ~60 MB/s read-from-disk requirement, which is probably impossible / pushing it. Obviously we'd have to do some level of image compression (perhaps native openGL/es compression formats via PVRTC)... but that's kind crazy. Maybe there is a library out there that I can use? Idea 7: Combine everything into a single movie asset, and seekToTime One idea that might be easier than some of the above, is to combine everything into a single movie, and use seekToTime. The thing is that we'd be jumping all around the place. Essentially random access into the movie. I think this may actually work out okay: avplayer-movie-playing-lag-in-ios5 Which approach do you think would be best? So far, I've not made that much progress in terms of reducing the lag.

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  • F# Static Member Type Constraints

    - by Stephen Swensen
    I'm trying to define a function, factorize, which uses structural type constraints (requires static members Zero, One, +, and /) similar to Seq.sum so that it can be used with int, long, bigint, etc. I can't seem to get the syntax right, and can't find a lot of resources on the subject. This is what I have, please help. let inline factorize (n:^NUM) = ^NUM : (static member get_Zero: unit->(^NUM)) ^NUM : (static member get_One: unit->(^NUM)) let rec factorize (n:^NUM) (j:^NUM) (flist: ^NUM list) = if n = ^NUM.One then flist elif n % j = ^NUM.Zero then factorize (n/j) (^NUM.One + ^NUM.One) (j::flist) else factorize n (j + ^NUM.One) (flist) factorize n (^NUM.One + ^NUM.One) []

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  • Javascript replace last occurence of text in a string

    - by Ruth
    Hi all see my code snippet below: var list = ['one', 'two', 'three', 'four']; var str = 'one two, one three, one four, one]; for ( var i = 0; i < list.length; i++) { if (str.endsWith(list[i]) { str = str.replace(list[i], 'finsih') } } I want to replace the last occurence of the word one with the word finish in the string, what I have will not work because the replace method will only replace the first occurence of it. Does anyone know how I can amend that snippet so that it only replaces the last instance of 'one' Thank you Ruth

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  • compiling actionscript from command line using MXMLC

    - by I. J. Kennedy
    I have a tiny actionscript "project" consisting of two files, call them foo.as and bar.as. For reasons I won't go into, I really really want to build the .SWF from the command line, without setting up a formal project of any kind. Every compiler I've ever used lets you do this, but for the life of me I can't figure out how to coerce MXMLC into compiling these two files and linking them into a SWF. Naively, I try MXMLC foo.as bar.as but I'm informed that only one source file is allowed. Ok, supposing I compiled these two files separately, how would I link them together to get the final SWF? NOTE: The only reason I have two files instead of one is the requirement of only one class per file. I tried putting both classes in one file and making one of the classes "private" or "internal" but neither of these ideas worked. I would be ecstatic to find out I can put more than one class in a file (with only one being public).

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  • Constructor overriding

    - by demas
    I have a library with a class: class One def initialize puts "one initialize" end end I can not change the declaration and difinition of this class. I need create new class with my own constructor. Like this: class Two < One def initialize(some) puts some super end end one = One.new one = Two.new("thing") But when I launch code I got error: [[email protected]][~/temp]% ruby test.rb one initialize thing test.rb:10:in `initialize': wrong number of arguments (1 for 0) (ArgumentError) from test.rb:15:in `new' from test.rb:15:in `<main>'

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  • How to convert JavaScript dictionary into Python syntax

    - by Sputnix
    Writing out javascript dictionary from inside of JavaScript- enabled application (such as Adobe) into external .jsx file (or any other .txt file) the context of resulted file dictionary looks like: ({one:"1", two:"2"}) (Please note that each dictionary keys are written as they are the variables name (which is not true). A next step is to read this .jsx file with Python. I need to find a way to convert ({one:"1", two:"2"}) into Python dictionary syntax such as: {'one':"1", 'two':"2"} It has been already suggested that instead of using JavaScript's built-in dict.toSource() it would make more sense to use JSON which would write a dictionary content in similar to Python syntax. But unfortunately using JSON is not an option for me. I need to find a way to convert ({one:"1", two:"2"}) into {'one':"1", 'two':"2"} using Python alone. Any suggestions on how to achieve it? Once again, the problem mostly in dictionary keys syntax which inside of Python look like variable names instead of strings-like dictionary keys names: one vs "one"

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