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  • Some clarification on rvalue references

    - by Dennis Zickefoose
    First: where are std::move and std::forward defined? I know what they do, but I can't find proof that any standard header is required to include them. In gcc44 sometimes std::move is available, and sometimes its not, so a definitive include directive would be useful. When implementing move semantics, the source is presumably left in an undefined state. Should this state necessarily be a valid state for the object? Obviously, you need to be able to call the object's destructor, and be able to assign to it by whatever means the class exposes. But should other operations be valid? I suppose what I'm asking is, if your class guarantees certain invariants, should you strive to enforce those invariants when the user has said they don't care about them anymore? Next: when you don't care about move semantics, are there any limitations that would cause a non-const reference to be preferred over an rvalue reference when dealing with function parameters? void function(T&); over void function(T&&); From a caller's perspective, being able to pass functions temporary values is occasionally useful, so it seems as though one should grant that option whenever it is feasible to do so. And rvalue references are themselves lvalues, so you can't inadvertently call a move-constructor instead of a copy-constructor, or something like that. I don't see a downside, but I'm sure there is one. Which brings me to my final question. You still can not bind temporaries to non-const references. But you can bind them to non-const rvalue references. And you can then pass along that reference as a non-const reference in another function. void function1(int& r) { r++; } void function2(int&& r) { function1(r); } int main() { function1(5); //bad function2(5); //good } Besides the fact that it doesn't do anything, is there anything wrong with that code? My gut says of course not, since changing rvalue references is kind of the whole point to their existence. And if the passed value is legitimately const, the compiler will catch it and yell at you. But by all appearances, this is a runaround of a mechanism that was presumably put in place for a reason, so I'd just like confirmation that I'm not doing anything foolish.

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  • Looking for an Open Source Project in need of help

    - by hvidgaard
    Hi StackOverflow! I'm a CS student on well on my way to graduate. I have had a difficult time of finding relevant student jobs (they seems to be taken merely hours after the notice gets on the board) , so instead I'm looking for an open source project in need of help. I'm aware that I should choose one that I use, but I'm not aware of any OS-project that I use that needs help. That's why I'm asking you. I don't have any deep experience, but I here are some of my biggest projects so far: BitTorrent-ish client in Python (a subset of BitTorrent) HTTP 1.1 webserver in Java Compiler from a subset of Java to run on JRE Flash-framework project to model an iPad look and feel (not to run actual iPad programs) complete with an API for programs. Complete MySQL database for a booking system, with departure and arrival times, so you could only book valid tickets (with a Java frontend). I know, Java and languages like AS3 and C# feels natural per se, Python, and have done a fair bit of hacking around in C, but I don't feel very comfortable with it. Mostly I'm afraid to make a fuckup because I have such a high degree of control. I would like to think I'm well aware of good software design practices, but in reality what I do is ask myself "would I like to use/maintain this?", and I love to refactor my code because I see optimizations. I love algorithms and to make them run in the best possible time. I don't have any preferred domain to work in, but I wouldn't mind it to be graphics or math heavy. Ideally I'm looking for a project in C++ to learn the in's and out's of it, but I'm well aware that I don't know that language very well. I would like to have a mentor-like figure until I'm confident enough to stand on my own, not one to review all my code (I'm sure someone will to start with anyway), but to ask questions about the project and language in question. I do have a wife and two children, so don't expect me to put in 10+ hours every week. In return I can work on my own, I strive to program modular and maintainable code. Know how to read an API, use Google, StackOverflow and online resources in general. If you have any questions, shoot. I'm looking forward to your suggestions.

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  • functional dependencies vs type families

    - by mhwombat
    I'm developing a framework for running experiments with artificial life, and I'm trying to use type families instead of functional dependencies. Type families seems to be the preferred approach among Haskellers, but I've run into a situation where functional dependencies seem like a better fit. Am I missing a trick? Here's the design using type families. (This code compiles OK.) {-# LANGUAGE TypeFamilies, FlexibleContexts #-} import Control.Monad.State (StateT) class Agent a where agentId :: a -> String liveALittle :: Universe u => a -> StateT u IO a -- plus other functions class Universe u where type MyAgent u :: * withAgent :: (MyAgent u -> StateT u IO (MyAgent u)) -> String -> StateT u IO () -- plus other functions data SimpleUniverse = SimpleUniverse { mainDir :: FilePath -- plus other fields } defaultWithAgent :: (MyAgent u -> StateT u IO (MyAgent u)) -> String -> StateT u IO () defaultWithAgent = undefined -- stub -- plus default implementations for other functions -- -- In order to use my framework, the user will need to create a typeclass -- that implements the Agent class... -- data Bug = Bug String deriving (Show, Eq) instance Agent Bug where agentId (Bug s) = s liveALittle bug = return bug -- stub -- -- .. and they'll also need to make SimpleUniverse an instance of Universe -- for their agent type. -- instance Universe SimpleUniverse where type MyAgent SimpleUniverse = Bug withAgent = defaultWithAgent -- boilerplate -- plus similar boilerplate for other functions Is there a way to avoid forcing my users to write those last two lines of boilerplate? Compare with the version using fundeps, below, which seems to make things simpler for my users. (The use of UndecideableInstances may be a red flag.) (This code also compiles OK.) {-# LANGUAGE MultiParamTypeClasses, FunctionalDependencies, FlexibleInstances, UndecidableInstances #-} import Control.Monad.State (StateT) class Agent a where agentId :: a -> String liveALittle :: Universe u a => a -> StateT u IO a -- plus other functions class Universe u a | u -> a where withAgent :: Agent a => (a -> StateT u IO a) -> String -> StateT u IO () -- plus other functions data SimpleUniverse = SimpleUniverse { mainDir :: FilePath -- plus other fields } instance Universe SimpleUniverse a where withAgent = undefined -- stub -- plus implementations for other functions -- -- In order to use my framework, the user will need to create a typeclass -- that implements the Agent class... -- data Bug = Bug String deriving (Show, Eq) instance Agent Bug where agentId (Bug s) = s liveALittle bug = return bug -- stub -- -- And now my users only have to write stuff like... -- u :: SimpleUniverse u = SimpleUniverse "mydir"

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  • How to created filtered reports in WPF?

    - by Michael Goyote
    Creating reports in WPF. I have two related tables. Table A-Customer: CustomerID(PK) Names Phone Number Customer Num Table B-Items: Products Price CustomerID I want to be able to generate a report like this: CustomerA Items Price Item A 10 Item B 10 Item C 10 --------------- Total 30 So this is what I have done: <Window x:Class="ReportViewerWPF.MainWindow" xmlns="http://schemas.microsoft.com/winfx/2006/xaml/presentation" xmlns:x="http://schemas.microsoft.com/winfx/2006/xaml" xmlns:rv="clr-namespace:Microsoft.Reporting.WinForms; assembly=Microsoft.ReportViewer.WinForms" Title="Customer Report" Height="300" Width="400"> <Grid> <WindowsFormsHost Name="windowsFormsHost1"> <rv:ReportViewer x:Name="reportViewer1"/> </WindowsFormsHost> </Grid> Then I created a dataset and loaded the two tables, followed by a report wizard (dragged all the available fields and dropped them to the Values pane). The code behind the WPF window is this: public partial class CustomerReport : Window { public CustomerReport() { InitializeComponent(); _reportViewer.Load += ReportViewer_Load; } private bool _isReportViewerLoaded; private void ReportViewer_Load(object sender, EventArgs e) { if (!_isReportViewerLoaded) { Microsoft.Reporting.WinForms.ReportDataSource reportDataSource1 = new Microsoft.Reporting.WinForms.ReportDataSource(); HM2DataSet dataset = new HM2DataSet(); dataset.BeginInit(); reportDataSource1.Name = "DataSet";//This is the dataset name reportDataSource1.Value = dataset.CustomerTable; this.reportViewer1.LocalReport.DataSources.Add(reportDataSource1); this.reportViewer1.LocalReport.ReportPath = "../../Report3.rdlc"; dataset.EndInit(); HM2DataSetTableAdapters.CustomerTableAdapter funcTableAdapter = new HM2DataSetTableAdapters.CustomerTableAdapter(); funcTableAdapter.ClearBeforeFill = true; funcTableAdapter.Fill(dataset.CustomerTable); _reportViewer.RefreshReport(); _isReportViewerLoaded = true; } } As you might have guessed this loaded this list of customer with items and price: Customer Items Price Customer A Items A 10 Customer A Items B 10 Customer B Items D 10 Customer B Items C 10 How can I fine-tune this report to look like the one above, where the user can filter the customer he wants displayed on the report? Thanks in advance for the help. I would have preferred to use LINQ whenever filtering data

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  • Need Google Map InfoWindow Hyperlink to Open Content in Overlay (Fusion Table Usage)

    - by McKev
    I have the following code established to render the map in my site. When the map is clicked, the info window pops up with a bunch of content including a hyperlink to open up a website with a form in it. I would like to utilize a function like fancybox to open up this link "form" in an overlay. I have read that fancybox doesn't support calling the function from within an iframe, and was wondering if there was a way to pass the link data to the DOM and trigger the fancybox (or another overlay option) in another way? Maybe a callback trick - any tips would be much appreciated! <style> #map-canvas { width:850px; height:600px; } </style> <script type="text/javascript" src="http://maps.google.com/maps/api/js?sensor=true"></script> <script src="http://gmaps-utility-gis.googlecode.com/svn/trunk/fusiontips/src/fusiontips.js" type="text/javascript"></script> <script type="text/javascript"> var map; var tableid = "1nDFsxuYxr54viD_fuH7fGm1QRZRdcxFKbSwwRjk"; var layer; var initialLocation; var browserSupportFlag = new Boolean(); var uscenter = new google.maps.LatLng(37.6970, -91.8096); function initialize() { map = new google.maps.Map(document.getElementById('map-canvas'), { zoom: 4, mapTypeId: google.maps.MapTypeId.ROADMAP }); layer = new google.maps.FusionTablesLayer({ query: { select: "'Geometry'", from: tableid }, map: map }); //http://gmaps-utility-gis.googlecode.com/svn/trunk/fusiontips/docs/reference.html layer.enableMapTips({ select: "'Contact Name','Contact Title','Contact Location','Contact Phone'", from: tableid, geometryColumn: 'Geometry', suppressMapTips: false, delay: 500, tolerance: 8 }); ; // Try W3C Geolocation (Preferred) if(navigator.geolocation) { browserSupportFlag = true; navigator.geolocation.getCurrentPosition(function(position) { initialLocation = new google.maps.LatLng(position.coords.latitude,position.coords.longitude); map.setCenter(initialLocation); //Custom Marker var pinColor = "A83C0A"; var pinImage = new google.maps.MarkerImage("http://chart.apis.google.com/chart?chst=d_map_pin_letter&chld=%E2%80%A2|" + pinColor, new google.maps.Size(21, 34), new google.maps.Point(0,0), new google.maps.Point(10, 34)); var pinShadow = new google.maps.MarkerImage("http://chart.apis.google.com/chart?chst=d_map_pin_shadow", new google.maps.Size(40, 37), new google.maps.Point(0, 0), new google.maps.Point(12, 35)); new google.maps.Marker({ position: initialLocation, map: map, icon: pinImage, shadow: pinShadow }); }, function() { handleNoGeolocation(browserSupportFlag); }); } // Browser doesn't support Geolocation else { browserSupportFlag = false; handleNoGeolocation(browserSupportFlag); } function handleNoGeolocation(errorFlag) { if (errorFlag == true) { //Geolocation service failed initialLocation = uscenter; } else { //Browser doesn't support geolocation initialLocation = uscenter; } map.setCenter(initialLocation); } } google.maps.event.addDomListener(window, 'load', initialize); </script>

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  • SQL Server architecture guidance

    - by Liam
    Hi, We are designing a new version of our existing product on a new schema. Its an internal web application with possibly 100 concurrent users (max)This will run on a SQL Server 2008 database. On of the discussion items recently is whether we should have a single database of split the database for performance reasons across 2 separate databases. The database could grow anywhere from 50-100GB over 5 years. We are Developers and not DBAs so it would be nice to get some general guidance. [I know the answer is not simple as it depends on the schema, archiving policy, amount of data etc. ] Option 1 Single Main Database [This is my preferred option]. The plan would be to have all the tables in a single database and possibly to use file groups and partitioning to separate the data if required across multiple disks. [Use schema if appropriate]. This should deal with the performance concerns One of the comments wrt this was that the a single server instance would still be processing this data so there would still be a processing bottle neck. For reporting we could have a separate reporting DB but this is still being discussed. Option 2 Split the database into 2 separate databases DB1 - Customers, Accounts, Customer resources etc DB2 - This would contain the bulk of the data [i.e. Vehicle tracking data, financial transaction tables etc]. These tables would typically contain a lot of data. [It could reside on a separate server if required] This plan would involve keeping the main data in a smaller database [DB1] and retaining the [mainly] read only transaction type data in a separate DB [DB2]. The UI would mainly read from DB1 and thus be more responsive. [I'm aware that this option makes it harder for Referential Integrity to be enforced.] Points for consideration As we are at the design stage we can at least make proper use of indexes to deal performance issues so thats why option 1 to me is attractive and its more of a standard approach. For both options we are considering implementing an archiving database. Apologies for the long Question. In summary the question is 1 DB or 2? Thanks in advance, Liam

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  • Loosely coupled implicit conversion

    - by ltjax
    Implicit conversion can be really useful when types are semantically equivalent. For example, imagine two libraries that implement a type identically, but in different namespaces. Or just a type that is mostly identical, except for some semantic-sugar here and there. Now you cannot pass one type into a function (in one of those libraries) that was designed to use the other, unless that function is a template. If it's not, you have to somehow convert one type into the other. This should be trivial (or otherwise the types are not so identical after-all!) but calling the conversion explicitly bloats your code with mostly meaningless function-calls. While such conversion functions might actually copy some values around, they essentially do nothing from a high-level "programmers" point-of-view. Implicit conversion constructors and operators could obviously help, but they introduce coupling, so that one of those types has to know about the other. Usually, at least when dealing with libraries, that is not the case, because the presence of one of those types makes the other one redundant. Also, you cannot always change libraries. Now I see two options on how to make implicit conversion work in user-code: The first would be to provide a proxy-type, that implements conversion-operators and conversion-constructors (and assignments) for all the involved types, and always use that. The second requires a minimal change to the libraries, but allows great flexibility: Add a conversion-constructor for each involved type that can be externally optionally enabled. For example, for a type A add a constructor: template <class T> A( const T& src, typename boost::enable_if<conversion_enabled<T,A>>::type* ignore=0 ) { *this = convert(src); } and a template template <class X, class Y> struct conversion_enabled : public boost::mpl::false_ {}; that disables the implicit conversion by default. Then to enable conversion between two types, specialize the template: template <> struct conversion_enabled<OtherA, A> : public boost::mpl::true_ {}; and implement a convert function that can be found through ADL. I would personally prefer to use the second variant, unless there are strong arguments against it. Now to the actual question(s): What's the preferred way to associate types for implicit conversion? Are my suggestions good ideas? Are there any downsides to either approach? Is allowing conversions like that dangerous? Should library implementers in-general supply the second method when it's likely that their type will be replicated in software they are most likely beeing used with (I'm thinking of 3d-rendering middle-ware here, where most of those packages implement a 3D vector).

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  • Project Euler #18 - how to brute force all possible paths in tree-like structure using Python?

    - by euler user
    Am trying to learn Python the Atlantic way and am stuck on Project Euler #18. All of the stuff I can find on the web (and there's a LOT more googling that happened beyond that) is some variation on 'well you COULD brute force it, but here's a more elegant solution'... I get it, I totally do. There are really neat solutions out there, and I look forward to the day where the phrase 'acyclic graph' conjures up something more than a hazy, 1 megapixel resolution in my head. But I need to walk before I run here, see the state, and toy around with the brute force answer. So, question: how do I generate (enumerate?) all valid paths for the triangle in Project Euler #18 and store them in an appropriate python data structure? (A list of lists is my initial inclination?). I don't want the answer - I want to know how to brute force all the paths and store them into a data structure. Here's what I've got. I'm definitely looping over the data set wrong. The desired behavior would be to go 'depth first(?)' rather than just looping over each row ineffectually.. I read ch. 3 of Norvig's book but couldn't translate the psuedo-code. Tried reading over the AIMA python library for ch. 3 but it makes too many leaps. triangle = [ [75], [95, 64], [17, 47, 82], [18, 35, 87, 10], [20, 4, 82, 47, 65], [19, 1, 23, 75, 3, 34], [88, 2, 77, 73, 7, 63, 67], [99, 65, 4, 28, 6, 16, 70, 92], [41, 41, 26, 56, 83, 40, 80, 70, 33], [41, 48, 72, 33, 47, 32, 37, 16, 94, 29], [53, 71, 44, 65, 25, 43, 91, 52, 97, 51, 14], [70, 11, 33, 28, 77, 73, 17, 78, 39, 68, 17, 57], [91, 71, 52, 38, 17, 14, 91, 43, 58, 50, 27, 29, 48], [63, 66, 4, 68, 89, 53, 67, 30, 73, 16, 69, 87, 40, 31], [04, 62, 98, 27, 23, 9, 70, 98, 73, 93, 38, 53, 60, 4, 23], ] def expand_node(r, c): return [[r+1,c+0],[r+1,c+1]] all_paths = [] my_path = [] for i in xrange(0, len(triangle)): for j in xrange(0, len(triangle[i])): print 'row ', i, ' and col ', j, ' value is ', triangle[i][j] ??my_path = somehow chain these together??? if my_path not in all_paths all_paths.append(my_path) Answers that avoid external libraries (like itertools) preferred.

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  • Which is the "best" data access framework/approach for C# and .NET?

    - by Frans
    (EDIT: I made it a community wiki as it is more suited to a collaborative format.) There are a plethora of ways to access SQL Server and other databases from .NET. All have their pros and cons and it will never be a simple question of which is "best" - the answer will always be "it depends". However, I am looking for a comparison at a high level of the different approaches and frameworks in the context of different levels of systems. For example, I would imagine that for a quick-and-dirty Web 2.0 application the answer would be very different from an in-house Enterprise-level CRUD application. I am aware that there are numerous questions on Stack Overflow dealing with subsets of this question, but I think it would be useful to try to build a summary comparison. I will endeavour to update the question with corrections and clarifications as we go. So far, this is my understanding at a high level - but I am sure it is wrong... I am primarily focusing on the Microsoft approaches to keep this focused. ADO.NET Entity Framework Database agnostic Good because it allows swapping backends in and out Bad because it can hit performance and database vendors are not too happy about it Seems to be MS's preferred route for the future Complicated to learn (though, see 267357) It is accessed through LINQ to Entities so provides ORM, thus allowing abstraction in your code LINQ to SQL Uncertain future (see Is LINQ to SQL truly dead?) Easy to learn (?) Only works with MS SQL Server See also Pros and cons of LINQ "Standard" ADO.NET No ORM No abstraction so you are back to "roll your own" and play with dynamically generated SQL Direct access, allows potentially better performance This ties in to the age-old debate of whether to focus on objects or relational data, to which the answer of course is "it depends on where the bulk of the work is" and since that is an unanswerable question hopefully we don't have to go in to that too much. IMHO, if your application is primarily manipulating large amounts of data, it does not make sense to abstract it too much into objects in the front-end code, you are better off using stored procedures and dynamic SQL to do as much of the work as possible on the back-end. Whereas, if you primarily have user interaction which causes database interaction at the level of tens or hundreds of rows then ORM makes complete sense. So, I guess my argument for good old-fashioned ADO.NET would be in the case where you manipulate and modify large datasets, in which case you will benefit from the direct access to the backend. Another case, of course, is where you have to access a legacy database that is already guarded by stored procedures. ASP.NET Data Source Controls Are these something altogether different or just a layer over standard ADO.NET? - Would you really use these if you had a DAL or if you implemented LINQ or Entities? NHibernate Seems to be a very powerful and powerful ORM? Open source Some other relevant links; NHibernate or LINQ to SQL Entity Framework vs LINQ to SQL

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  • C - What is the proper format to allow a function to show an error was encountered?

    - by BrainSteel
    I have a question about what a function should do if the arguments to said function don't line up quite right, through no fault of the function call. Since that sentence doesn't make much sense, I'll offer my current issue. To keep it simple, here is the most relevant and basic function I have. float getYValueAt(float x, PHYS_Line line, unsigned short* error) *error = 0; if(x < line.start.x || x > line.end.x){ *error = 1; return -1; } if(line.slope.value != 0){ //line's equation: y - line.start.y = line.slope.value(x - line.start.x) return line.slope.value * (x - line.start.x) + line.start.y; } else if(line.slope.denom == 0){ if(line.start.x == x) return line.start.y; else{ *error = 1; return -1; } } else if(line.slope.num == 0){ return line.start.y; } } The function attempts to find the point on a line, given a certain x value. However, under some circumstances, this may not be possible. For example, on the line x = 3, if 5 is passed as a value, we would have a problem. Another problem arises if the chosen x value is not within the interval the line is on. For this, I included the error pointer. Given this format, a function call could work as follows: void foo(PHYS_Line some_line){ unsigned short error = 0; float y = getYValueAt(5, some_line, &error); if(error) fooey(); else do_something_with_y(y); } My question pertains to the error. Note that the value returned is allowed to be negative. Returning -1 does not ensure that an error has occurred. I know that it is sometimes preferred to use the following method to track an error: float* getYValueAt(float x, PHYS_Line line); and then return NULL if an error occurs, but I believe this requires dynamic memory allocation, which seems even less sightly than the solution I was using. So, what is standard practice for an error occurring?

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  • c++ std::ostringstream vs std::string::append

    - by NickSoft
    In all examples that use some kind of buffering I see they use stream instead of string. How is std::ostringstream and << operator different than using string.append. Which one is faster and which one uses less resourses (memory). One difference I know is that you can output different types into output stream (like integer) rather than the limited types that string::append accepts. Here is an example: std::ostringstream os; os << "Content-Type: " << contentType << ";charset=" << charset << "\r\n"; std::string header = os.str(); vs std::string header("Content-Type: "); header.append(contentType); header.append(";charset="); header.append(charset); header.append("\r\n"); Obviously using stream is shorter, but I think append returns reference to the string so it can be written like this: std::string header("Content-Type: "); header.append(contentType) .append(";charset=") .append(charset) .append("\r\n"); And with output stream you can do: std::string content; ... os << "Content-Length: " << content.length() << "\r\n"; But what about memory usage and speed? Especially when used in a big loop. Update: To be more clear the question is: Which one should I use and why? Is there situations when one is preferred or the other? For performance and memory ... well I think benchmark is the only way since every implementation could be different. Update 2: Well I don't get clear idea what should I use from the answers which means that any of them will do the job, plus vector. Cubbi did nice benchmark with the addition of Dietmar Kühl that the biggest difference is construction of those objects. If you are looking for an answer you should check that too. I'll wait a bit more for other answers (look previous update) and if I don't get one I think I'll accept Tolga's answer because his suggestion to use vector is already done before which means vector should be less resource hungry.

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  • Using images in QListWidget, is this possible?

    - by Neko
    Hi, again. :) As with the last few times I posted here, I have a question about QT, which I am using to create a chat messenger client. To display the list of online users, I'm using a QListWidget, as created like this: listWidget = new QListWidget(horizontalLayoutWidget); listWidget->setObjectName("userList"); QSizePolicy sizePolicy1(QSizePolicy::Preferred, QSizePolicy::Expanding); sizePolicy1.setHorizontalStretch(0); sizePolicy1.setVerticalStretch(0); sizePolicy1.setHeightForWidth(listWidget->sizePolicy().hasHeightForWidth()); listWidget->setSizePolicy(sizePolicy1); listWidget->setMinimumSize(QSize(30, 0)); listWidget->setMaximumSize(QSize(150, 16777215)); listWidget->setBaseSize(QSize(100, 0)); listWidget->setContextMenuPolicy(Qt::CustomContextMenu); Users are shown by constantly refreshing the list, like this: (Note: There are different channels, with different userlists, so refreshing it is the most efficient thing to do, as far as I know.) void FMessenger::refreshUserlist() { if (currentPanel == 0) return; listWidget = this->findChild<QListWidget *>(QString("userList")); listWidget->clear(); QList<FCharacter*> charList = currentPanel->charList(); QListWidgetItem* charitem = 0; FCharacter* character; foreach(character, charList) { charitem = new QListWidgetItem(character->name()); // charitem->setIcon(QIcon(":/Images/status.png")); listWidget->addItem(charitem); } } This has always worked perfectly. The line that I commented out is the one I have problems with: my current goal is to be able to display a user's online status with an image, which represents whether they are busy, away, available, etc. Using setIcon() does absolutely nothing though, apparently; the items still show up as they used to, without icons. I'm aware that this is probably not the way this function needs to be used, but I have found little documentation about it online, and absolutely no useful examples of implementations. My question is, can anybody help me with fixing this problem? Any help would be appreciated terrifically, like always!

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  • How do I force a Coldfusion cfc to output numeric data over JSON as a string?

    - by Dan Sorensen
    I'm calling a Coldfusion component (cfc) using jQuery.post(). I need an integer or string representation of the number returned for use in a URL. {"PAGE":"My Page Title","ID":19382} or {"PAGE":"My Page Title","ID":"19382"} Instead what I get back is a decimal: {"PAGE":"My Page Title","ID":19382.0} Needed to update the following HTML: <a href="page.cfm?id=19382" id="pagelink">My Page Title</a> Conceptually, I suppose there are multiple answers: 1) I could use jQuery to grab the number left of the decimal point. 2) I could force Coldfusion to send the number as a string. 3) I could generate the whole link server side and just replace the whole link tag HTML (not the preferred answer, but maybe it is the best) Does anyone know how to do 1 or 2? Is 3 better? Relevant Javascript: (Not optimized) $(".link").live('click', function () { var $linkID, serviceUrl; serviceUrl = "mycfc.cfc?method=getPage"; $linkID = $(this).attr("rel"); $.post(serviceUrl, { linkid: $linkID }, function (result) { $('#pagelink').val(result.TITLE); if (result.FMKEY.length) { // NEED the ID number WITHOUT the .0 at the end $('#pagelink').attr("href") = "page.cfm?id=" + result.ID; $('#pagelink').text(result.TITLE); } }, "json"); }); My CFC: <component output="no"> <cfsetting showdebugoutput="no"> <cffunction name="getPage" access="remote" returnFormat="JSON" output="no" hint="Looks up a Page Title and ID"> <cfargument name="linkID" type="string" required="yes"> <cfset var page = queryNew("id,title")> <cfset var result = structNew()> <cfquery datasource="myDatasource" name="page"> SELECT TOP 1 id, title FROM pages WHERE linkID = <cfqueryparam cfsqltype="cf_sql_integer" value="#arguments.linkID#"> </cfquery> <cfif page.recordcount> <cfset result.id = page.id> <cfset result.title = page.title> </cfif> <cfreturn result> </cffunction> </component>

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  • Planning to create PDF files in Ruby on Rails

    - by deau
    Hi there, A Ruby on Rails app will have access to a number of images and fonts. The images are components of a visual layout which will be stored separately as a set of rules. The rules specify document dimensions along with which images are used and where. The app needs to take these rules, fetch the images, and generate a PDF that is ready for local printing or emailing. The fonts will also be important. The user needs to customize the layout by inputting text which will be included in the PDF. The PDF must therefore also contain the desired font so that the document renders identically across different machines. Each PDF may have many pages. Each page may have different dimensions but this is not essential. Either way, the ability to manipulate the dimensions and margins given by the PDF is essential. The only thing that needs to be regularly changed is the text. If this is takes too much development then the app can store the layouts in 3rd party PDFs and edit the textual content directly. Eventually though, this will prove too restrictive on the apps intended functionality so I would prefer the app to generate the PDF's itself. I have never worked with PDFs before and, for the most part, I've never had to output anything to the user outside their monitor. A printed medium could require a very different approach to get the best results. If anyone has any advice on how to model the PDF format this it would be really appreciated. The technical aspects of printing such as bleed, resolution and colour have already been factored in to the layouts and images. I am aware that PDF is a proprietary file format and I want to use free or open source software. I have seen a number of Ruby libraries for generating PDF files but because I am new on this scene I have no way to reliably compare them and too little time to implement and test them all. I also have the option of using C to handle this feature and if this is process intensive then that might be preferred. What should I be thinking about and how should I be planning to implement this?

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  • Passing arguments between classes - use public properties or pass a properties class as argument?

    - by devoured elysium
    So let's assume I have a class named ABC that will have a list of Point objects. I need to make some drawing logic with them. Each one of those Point objects will have a Draw() method that will be called by the ABC class. The Draw() method code will need info from ABC class. I can only see two ways to make them have this info: Having Abc class make public some properties that would allow draw() to make its decisions. Having Abc class pass to draw() a class full of properties. The properties in both cases would be the same, my question is what is preferred in this case. Maybe the second approach is more flexible? Maybe not? I don't see here a clear winner, but that sure has more to do with my inexperience than any other thing. If there are other good approaches, feel free to share them. Here are both cases: class Abc1 { public property a; public property b; public property c; ... public property z; public void method1(); ... public void methodn(); } and here is approach 2: class Abc2 { //here we make take down all properties public void method1(); ... public void methodn(); } class Abc2MethodArgs { //and we put them here. this class will be passed as argument to //Point's draw() method! public property a; public property b; public property c; ... public property z; } Also, if there are any "formal" names for these two approaches, I'd like to know them so I can better choose the tags/thread name, so it's more useful for searching purposes. That or feel free to edit them.

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  • Google Map lng + lat to hidden field not working

    - by user547794
    Hello, I am trying to get Marker data into hidden fields on my form. I'm not sure why this isn't working, it must be something in my js syntax: var initialLocation; var siberia = new google.maps.LatLng(60, 105); var newyork = new google.maps.LatLng(40.69847032728747, -73.9514422416687); var browserSupportFlag = new Boolean(); function initialize() { var myOptions = { zoom: 6, mapTypeId: google.maps.MapTypeId.HYBRID }; var map = new google.maps.Map(document.getElementById("map_canvas"), myOptions); myListener = google.maps.event.addListener(map, 'click', function(event) { placeMarker(event.latLng), google.maps.event.removeListener(myListener); }); // Try W3C Geolocation (Preferred) if(navigator.geolocation) { browserSupportFlag = true; navigator.geolocation.getCurrentPosition(function(position) { initialLocation = new google.maps.LatLng(position.coords.latitude,position.coords.longitude); map.setCenter(initialLocation); }, function() { handleNoGeolocation(browserSupportFlag); }); // Try Google Gears Geolocation } else if (google.gears) { browserSupportFlag = true; var geo = google.gears.factory.create('beta.geolocation'); geo.getCurrentPosition(function(position) { initialLocation = new google.maps.LatLng(position.latitude,position.longitude); map.setCenter(initialLocation); }, function() { handleNoGeoLocation(browserSupportFlag); }); // Browser doesn't support Geolocation } else { browserSupportFlag = false; handleNoGeolocation(browserSupportFlag); } function handleNoGeolocation(errorFlag) { if (errorFlag == true) { alert("Geolocation service failed."); initialLocation = newyork; } else { alert("Your browser doesn't support geolocation. We've placed you in Siberia."); initialLocation = siberia; } } function placeMarker(location) { var marker = new google.maps.Marker({ position: location, map: map, draggable: true }); map.setCenter(location); } } var lat = latlng.lat(); var lng = latlng.lng(); document.getElementById("t1").value=lat; document.getElementById("t2").value=lng; <input type="hidden" name="lat" id="t1"> <input type="hidden" name="long" id="t2">

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  • getting html tags from xml and echoing in php?

    - by Whitney Sarah Rogers
    I am trying to echo a result from xml into my html code form expedia. But I ran into a problem. There text is a little messed up: <areaInformation> Distances are calculated in a straight line from the property&apos;s location to the point of interest or attraction, and may not reflect actual travel distance. &lt;br /&gt;&lt;br /&gt; Distances are displayed to the nearest 0.1 mile and kilometre. &lt;p&gt;La Isla Shopping Mall - 0.5 km / 0.3 mi &lt;br /&gt;Yamil Lu&apos;um - 0.5 km / 0.3 mi &lt;br /&gt;Acuario Interactivo - 0.6 km / 0.3 mi &lt;br /&gt;Luxury Avenue - 1.5 km / 0.9 mi &lt;br /&gt;Cancun Golf Club at Pok Ta Pok - 2.2 km / 1.3 mi &lt;br /&gt;Nautilus Diving and Training Center - 2.6 km / 1.6 mi &lt;br /&gt;Cancun Convention Center - 2.8 km / 1.7 mi &lt;br /&gt;Plaza Caracol - 2.8 km / 1.8 mi &lt;br /&gt;Playa Tortuga - 3.1 km / 1.9 mi &lt;br /&gt;Aquaworld - 3.6 km / 2.2 mi &lt;br /&gt;Playa Langosta - 4.1 km / 2.6 mi &lt;br /&gt;Museo de Arte Popular Mexicano - 4.6 km / 2.9 mi &lt;br /&gt;Playa Linda - 5 km / 3.1 mi &lt;br /&gt;Playa Delfines - 6.1 km / 3.8 mi &lt;br /&gt;El Rey Ruins - 6.2 km / 3.8 mi &lt;br /&gt;&lt;/p&gt;&lt;p&gt;The preferred airport for ME Cancun - Complete ME All Inclusive is Cancun, Quintana Roo (CUN-Cancun Intl.) - 14.3 km / 8.9 mi. &lt;/p&gt; </areaInformation> And I echo it in php: <div id="hotelInfo"><?php echo $areaInfo ?></div> And of course I get this in the browser window: Distances are calculated in a straight line from the property's location to the point of interest or attraction, and may not reflect actual travel distance. <br /><br /> Distances are displayed to the nearest 0.1 mile and kilometre. <p>La Isla Shopping Mall - 0.5 km / 0.3 mi <br />Yamil Lu'um - 0.5 km / 0.3 mi <br />Acuario Interac etc. How can I fix this??? Any help would be greatly apreciated! Thanks!

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  • Is it a good or bad practice to call instance methods from a java constructor?

    - by Steve
    There are several different ways I can initialize complex objects (with injected dependencies and required set-up of injected members), are all seem reasonable, but have various advantages and disadvantages. I'll give a concrete example: final class MyClass { private final Dependency dependency; @Inject public MyClass(Dependency dependency) { this.dependency = dependency; dependency.addHandler(new Handler() { @Override void handle(int foo) { MyClass.this.doSomething(foo); } }); doSomething(0); } private void doSomething(int foo) { dependency.doSomethingElse(foo+1); } } As you can see, the constructor does 3 things, including calling an instance method. I've been told that calling instance methods from a constructor is unsafe because it circumvents the compiler's checks for uninitialized members. I.e. I could have called doSomething(0) before setting this.dependency, which would have compiled but not worked. What is the best way to refactor this? Make doSomething static and pass in the dependency explicitly? In my actual case I have three instance methods and three member fields that all depend on one another, so this seems like a lot of extra boilerplate to make all three of these static. Move the addHandler and doSomething into an @Inject public void init() method. While use with Guice will be transparent, it requires any manual construction to be sure to call init() or else the object won't be fully-functional if someone forgets. Also, this exposes more of the API, both of which seem like bad ideas. Wrap a nested class to keep the dependency to make sure it behaves properly without exposing additional API:class DependencyManager { private final Dependency dependency; public DependecyManager(Dependency dependency) { ... } public doSomething(int foo) { ... } } @Inject public MyClass(Dependency dependency) { DependencyManager manager = new DependencyManager(dependency); manager.doSomething(0); } This pulls instance methods out of all constructors, but generates an extra layer of classes, and when I already had inner and anonymous classes (e.g. that handler) it can become confusing - when I tried this I was told to move the DependencyManager to a separate file, which is also distasteful because it's now multiple files to do a single thing. So what is the preferred way to deal with this sort of situation?

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  • How to calculate where bullet hits

    - by lkjoel
    I have been trying to write an FPS in C/X11/OpenGL, but the issue that I have encountered is with calculating where the bullet hits. I have used a horrible technique, and it only sometimes works: pos size, p; size.x = 0.1; size.z = 0.1; // Since the game is technically top-down (but in a 3D perspective) // Positions are in X/Z, no Y float f; // Counter float d = FIRE_MAX + 1 /* Shortest Distance */, d1 /* Distance being calculated */; x = 0; // Index of object to hit for (f = 0.0; f < FIRE_MAX; f += .01) { // Go forwards p.x = player->pos.x + f * sin(toRadians(player->rot.x)); p.z = player->pos.z - f * cos(toRadians(player->rot.x)); // Get all objects that collide with the current position of the bullet short* objs = _colDetectGetObjects(p, size, objects); for (i = 0; i < MAX_OBJECTS; i++) { if (objs[i] == -1) { continue; } // Check the distance between the object and the player d1 = sqrt( pow((objects[i].pos.x - player->pos.x), 2) + pow((objects[i].pos.z - player->pos.z), 2)); // If it's closer, set it as the object to hit if (d1 < d) { x = i; d = d1; } } // If there was an object, hit it if (x > 0) { hit(&objects[x], FIRE_DAMAGE, explosions, currtime); break; } } It just works by making a for-loop and calculating any objects that might collide with where the bullet currently is. This, of course, is very slow, and sometimes doesn't even work. What would be the preferred way to calculate where the bullet hits? I have thought of making a line and seeing if any objects collide with that line, but I have no idea how to do that kind of collision detection. EDIT: I guess my question is this: How do I calculate the nearest object colliding in a line (that might not be a straight 45/90 degree angle)? Or are there any simpler methods of calculating where the bullet hits? The bullet is sort of like a laser, in the sense that gravity does not affect it (writing an old-school game, so I don't want it to be too realistic)

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  • Name hiding from inherited classes

    - by Mercerbearman
    I have the following sample code and I wanted to know the correct way to get access to the Pass method in the CBar class. Currently I have found 3 ways to get access to this method and they are as follows: Casting the object, ((CBar *) &foo)-Pass(1, 2, 3); Using this syntax, foo.CBar::Pass(1,2,3); Use the "using" syntax in the CFoo class declaration, using CBar::Pass. The following is an example of a simple project to test this capability. Foo.h #include "bar.h" class CFoo : public CBar { private: double m_a; double m_b; public: CFoo(void); ~CFoo(void); void Pass(double a, double b); }; Foo.cpp #include "Foo.h" CFoo::CFoo(void) { m_a = 0.0; m_b = 0.0; } CFoo::~CFoo(void) { } void CFoo::Pass(double a, double b) { m_a = a; m_b = b; } Bar.h class CBar { int m_x; int m_y; int m_z; public: CBar(void); ~CBar(void); void Pass(int x, int y, int z); }; Bar.cpp #include "Bar.h" CBar::CBar(void) { m_x = 0; m_y = 0; m_z = 0; } CBar::~CBar(void) { } void CBar::Pass(int x, int y, int z) { m_x = x; m_y = y; m_z = z; } And my main class DoStuff.cpp #include "DoStuff.h" #include "Foo.h" CDoStuff::CDoStuff(void) { } CDoStuff::~CDoStuff(void) { } int main() { CFoo foo, foo1, foo2; //This only gets to the Pass method in Foo. foo.Pass(2.5, 3.5); //Gets access to Pass method in Bar. foo1.CBar::Pass(5,10,15); //Can also case and access by location for the same result?? ((CBar *) &foo2)->Pass(100,200,300); return 0; } Are each of these options viable? Are some preferred? Are there pitfalls with using any one of the methods listed? I am especially curious about the foo.CBar::Pass(1,2,3) syntax. Thanks, B

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  • How do I stop and repair a RAID 5 array that has failed and has I/O pending?

    - by Ben Hymers
    The short version: I have a failed RAID 5 array which has a bunch of processes hung waiting on I/O operations on it; how can I recover from this? The long version: Yesterday I noticed Samba access was being very sporadic; accessing the server's shares from Windows would randomly lock up explorer completely after clicking on one or two directories. I assumed it was Windows being a pain and left it. Today the problem is the same, so I did a little digging; the first thing I noticed was that running ps aux | grep smbd gives a lot of lines like this: ben 969 0.0 0.2 96088 4128 ? D 18:21 0:00 smbd -F root 1708 0.0 0.2 93468 4748 ? Ss 18:44 0:00 smbd -F root 1711 0.0 0.0 93468 1364 ? S 18:44 0:00 smbd -F ben 3148 0.0 0.2 96052 4160 ? D Mar07 0:00 smbd -F ... There are a lot of processes stuck in the "D" state. Running ps aux | grep " D" shows up some other processes including my nightly backup script, all of which need to access the volume mounted on my RAID array at some point. After some googling, I found that it might be down to the RAID array failing, so I checked /proc/mdstat, which shows this: ben@jack:~$ cat /proc/mdstat Personalities : [linear] [multipath] [raid0] [raid1] [raid6] [raid5] [raid4] [raid10] md0 : active raid5 sdb1[3](F) sdc1[1] sdd1[2] 2930271872 blocks level 5, 64k chunk, algorithm 2 [3/2] [_UU] unused devices: <none> And running mdadm --detail /dev/md0 gives this: ben@jack:~$ sudo mdadm --detail /dev/md0 /dev/md0: Version : 00.90 Creation Time : Sat Oct 31 20:53:10 2009 Raid Level : raid5 Array Size : 2930271872 (2794.53 GiB 3000.60 GB) Used Dev Size : 1465135936 (1397.26 GiB 1500.30 GB) Raid Devices : 3 Total Devices : 3 Preferred Minor : 0 Persistence : Superblock is persistent Update Time : Mon Mar 7 03:06:35 2011 State : active, degraded Active Devices : 2 Working Devices : 2 Failed Devices : 1 Spare Devices : 0 Layout : left-symmetric Chunk Size : 64K UUID : f114711a:c770de54:c8276759:b34deaa0 Events : 0.208245 Number Major Minor RaidDevice State 3 8 17 0 faulty spare rebuilding /dev/sdb1 1 8 33 1 active sync /dev/sdc1 2 8 49 2 active sync /dev/sdd1 I believe this says that sdb1 has failed, and so the array is running with two drives out of three 'up'. Some advice I found said to check /var/log/messages for notices of failures, and sure enough there are plenty: ben@jack:~$ grep sdb /var/log/messages ... Mar 7 03:06:35 jack kernel: [4525155.384937] md/raid:md0: read error NOT corrected!! (sector 400644912 on sdb1). Mar 7 03:06:35 jack kernel: [4525155.389686] md/raid:md0: read error not correctable (sector 400644920 on sdb1). Mar 7 03:06:35 jack kernel: [4525155.389686] md/raid:md0: read error not correctable (sector 400644928 on sdb1). Mar 7 03:06:35 jack kernel: [4525155.389688] md/raid:md0: read error not correctable (sector 400644936 on sdb1). Mar 7 03:06:56 jack kernel: [4525176.231603] sd 0:0:1:0: [sdb] Unhandled sense code Mar 7 03:06:56 jack kernel: [4525176.231605] sd 0:0:1:0: [sdb] Result: hostbyte=DID_OK driverbyte=DRIVER_SENSE Mar 7 03:06:56 jack kernel: [4525176.231608] sd 0:0:1:0: [sdb] Sense Key : Medium Error [current] [descriptor] Mar 7 03:06:56 jack kernel: [4525176.231623] sd 0:0:1:0: [sdb] Add. Sense: Unrecovered read error - auto reallocate failed Mar 7 03:06:56 jack kernel: [4525176.231627] sd 0:0:1:0: [sdb] CDB: Read(10): 28 00 17 e1 5f bf 00 01 00 00 To me it is clear that device sdb has failed, and I need to stop the array, shutdown, replace it, reboot, then repair the array, bring it back up and mount the filesystem. I cannot hot-swap a replacement drive in, and don't want to leave the array running in a degraded state. I believe I am supposed to unmount the filesystem before stopping the array, but that is failing, and that is where I'm stuck now: ben@jack:~$ sudo umount /storage umount: /storage: device is busy. (In some cases useful info about processes that use the device is found by lsof(8) or fuser(1)) It is indeed busy; there are some 30 or 40 processes waiting on I/O. What should I do? Should I kill all these processes and try again? Is that a wise move when they are 'uninterruptable'? What would happen if I tried to reboot? Please let me know what you think I should do. And please ask if you need any extra information to diagnose the problem or to help!

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  • Sign an OpenSSL .CSR with Microsoft Certificate Authority

    - by kce
    I'm in the process of building a Debian FreeRadius server that does 802.1x authentication for domain members. I would like to sign my radius server's SSL certificate (used for EAP-TLS) and leverage the domain's existing PKI. The radius server is joined to domain via Samba and has a machine account as displayed in Active Directory Users and Computers. The domain controller I'm trying to sign my radius server's key against does not have IIS installed so I can't use the preferred Certsrv webpage to generate the certificate. The MMC tools won't work as it can't access the certificate stores on the radius server because they don't exist. This leaves the certreq.exe utility. I'm generating my .CSR with the following command: openssl req -nodes -newkey rsa:1024 -keyout server.key -out server.csr The resulting .CSR: ******@mis-ke-lnx:~/G$ openssl req -text -noout -in mis-radius-lnx.csr Certificate Request: Data: Version: 0 (0x0) Subject: C=US, ST=Alaska, L=CITY, O=ORG, OU=DEPT, CN=ME/emailAddress=MYEMAIL Subject Public Key Info: Public Key Algorithm: rsaEncryption RSA Public Key: (1024 bit) Modulus (1024 bit): 00:a8:b3:0d:4b:3f:fa:a4:5f:78:0c:24:24:23:ac: cf:c5:28:af:af:a2:9b:07:23:67:4c:77:b5:e8:8a: 08:2e:c5:a3:37:e1:05:53:41:f3:4b:e1:56:44:d2: 27:c6:90:df:ae:3b:79:e4:20:c2:e4:d1:3e:22:df: 03:60:08:b7:f0:6b:39:4d:b4:5e:15:f7:1d:90:e8: 46:10:28:38:6a:62:c2:39:80:5a:92:73:37:85:37: d3:3e:57:55:b8:93:a3:43:ac:2b:de:0f:f8:ab:44: 13:8e:48:29:d7:8d:ce:e2:1d:2a:b7:2b:9d:88:ea: 79:64:3f:9a:7b:90:13:87:63 Exponent: 65537 (0x10001) Attributes: a0:00 Signature Algorithm: sha1WithRSAEncryption 35:57:3a:ec:82:fc:0a:8b:90:9a:11:6b:56:e7:a8:e4:91:df: 73:1a:59:d6:5f:90:07:83:46:aa:55:54:1c:f9:28:3e:a6:42: 48:0d:6b:da:58:e4:f5:7f:81:ee:e2:66:71:78:85:bd:7f:6d: 02:b6:9c:32:ad:fa:1f:53:0a:b4:38:25:65:c2:e4:37:00:16: 53:d2:da:f2:ad:cb:92:2b:58:15:f4:ea:02:1c:a3:1c:1f:59: 4b:0f:6c:53:70:ef:47:60:b6:87:c7:2c:39:85:d8:54:84:a1: b4:67:f0:d3:32:f4:8e:b3:76:04:a8:65:48:58:ad:3a:d2:c9: 3d:63 I'm trying to submit my certificate using the following certreq.exe command: certreq -submit -attrib "CertificateTemplate:Machine" server.csr I receive the following error upon doing so: RequestId: 601 Certificate not issued (Denied) Denied by Policy Module The DNS name is unavailable and cannot be added to the Subject Alternate name. 0x8009480f (-2146875377) Certificate Request Processor: The DNS name is unavailable and cannot be added to the Subject Alternate name. 0x8009480f (-2146875377) Denied by Policy Module My certificate authority has the following certificate templates available. If I try to submit by certreq.exe using "CertificiateTemplate:Computer" instead of "CertificateTemplate:Machine" I get an error reporting that "the requested certificate template is not supported by this CA." My google-foo has failed me so far on trying to understand this error... I feel like this should be a relatively simple task as X.509 is X.509 and OpenSSL generates the .CSRs in the required PKCS10 format. I can't be only one out there trying to sign a OpenSSL generated key on a Linux box with a Windows Certificate Authority, so how do I do this (perferably using the off-line certreq.exe tool)?

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  • What free space thresholds/limits are advisable for 640 GB and 2 TB hard disk drives with ZEVO ZFS on OS X?

    - by Graham Perrin
    Assuming that free space advice for ZEVO will not differ from advice for other modern implementations of ZFS … Question Please, what percentages or amounts of free space are advisable for hard disk drives of the following sizes? 640 GB 2 TB Thoughts A standard answer for modern implementations of ZFS might be "no more than 96 percent full". However if apply that to (say) a single-disk 640 GB dataset where some of the files most commonly used (by VirtualBox) are larger than 15 GB each, then I guess that blocks for those files will become sub optimally spread across the platters with around 26 GB free. I read that in most cases, fragmentation and defragmentation should not be a concern with ZFS. Sill, I like the mental picture of most fragments of a large .vdi in reasonably close proximity to each other. (Do features of ZFS make that wish for proximity too old-fashioned?) Side note: there might arise the question of how to optimise performance after a threshold is 'broken'. If it arises, I'll keep it separate. Background On a 640 GB StoreJet Transcend (product ID 0x2329) in the past I probably went beyond an advisable threshold. Currently the largest file is around 17 GB –  – and I doubt that any .vdi or other file on this disk will grow beyond 40 GB. (Ignore the purple masses, those are bundles of 8 MB band files.) Without HFS Plus: the thresholds of twenty, ten and five percent that I associate with Mobile Time Machine file system need not apply. I currently use ZEVO Community Edition 1.1.1 with Mountain Lion, OS X 10.8.2, but I'd like answers to be not too version-specific. References, chronological order ZFS Block Allocation (Jeff Bonwick's Blog) (2006-11-04) Space Maps (Jeff Bonwick's Blog) (2007-09-13) Doubling Exchange Performance (Bizarre ! Vous avez dit Bizarre ?) (2010-03-11) … So to solve this problem, what went in 2010/Q1 software release is multifold. The most important thing is: we increased the threshold at which we switched from 'first fit' (go fast) to 'best fit' (pack tight) from 70% full to 96% full. With TB drives, each slab is at least 5GB and 4% is still 200MB plenty of space and no need to do anything radical before that. This gave us the biggest bang. Second, instead of trying to reuse the same primary slabs until it failed an allocation we decided to stop giving the primary slab this preferential threatment as soon as the biggest allocation that could be satisfied by a slab was down to 128K (metaslab_df_alloc_threshold). At that point we were ready to switch to another slab that had more free space. We also decided to reduce the SMO bonus. Before, a slab that was 50% empty was preferred over slabs that had never been used. In order to foster more write aggregation, we reduced the threshold to 33% empty. This means that a random write workload now spread to more slabs where each one will have larger amount of free space leading to more write aggregation. Finally we also saw that slab loading was contributing to lower performance and implemented a slab prefetch mechanism to reduce down time associated with that operation. The conjunction of all these changes lead to 50% improved OLTP and 70% reduced variability from run to run … OLTP Improvements in Sun Storage 7000 2010.Q1 (Performance Profiles) (2010-03-11) Alasdair on Everything » ZFS runs really slowly when free disk usage goes above 80% (2010-07-18) where commentary includes: … OpenSolaris has changed this in onnv revision 11146 … [CFT] Improved ZFS metaslab code (faster write speed) (2010-08-22)

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  • Integrating HP Systems Insight Manager into an existing environment

    - by ewwhite
    I'm working with an environment that spans multiple data centers/sites and consists primarily of HP ProLiant servers (G5-G7) running Linux. The mix is 30% RHEL/CentOS, the rest are Gentoo :(. I also have a few dozen virtual machines running back-office and Windows servers on VMWare ESX hosts. I run OpenNMS to pull SNMP data from the various server nodes and networking devices. While OpenNMS works wonderfully for up/down, thresholds and notifications, it's native handling of traps is a little rough and the graphs are not particularly pretty. I use Orca/RRD graphs for performance trending and nice graphs. I'm tasked with inventorying the environment and wanted to come up with a clean way to organize server information. Since my environment is mostly HP, I've been playing with HP Systems Insight Manager as a way to extract server data and to deploy HP health/monitoring packages and firmware. The Gentoo systems eventually have to be converted to CentOS, so getting a quick assessment of what hardware is where would be great. Although I've read through a few hundred pages of HP manuals, I'm having a difficult time understanding how to get HP SIM to do what I want, though. My main problems are: I have about 40 subnets to deal with; 98% connected with private lines to facilities across the globe. I don't want to initiate an HP SIM discovery only to pull back every piece of intermediate networking hardware and equipment from all of the locations. I'd like this to focus on the servers. I have OpenNMS configured to accept traps. I don't want HP SIM to duplicate that effort. It seems like the built-in software deployment tool wants to overwrite the trapsink parameters for the systems it encounters during discovery. I have about 10 administrative username/password combinations in use across this infrastructure. Is there a more efficient way to get HP SIM to do the discovery or break discovery into manageable chunks? In terms of general workflow, do people typically install the HP Management Agents during the initial OS deployment (e.g. kickstart post script) or afterwards from HP SIM? Is HP SIM too thick/fat to be an inventory tool? I can't tell if it's meant to be used standalone or alongside other monitoring products. Since the majority of the systems I'm trying to track are those running Gentoo (in order to plan the move to CentOS), is there any way for HP SIM to extract system model information from them ( like dmidecode)? I have systems here where I may have an SSH key established, but not direct user or login access. Is there a way for me to import an SSH private/public key pair into HP SIM to reach out to the servers that can't accept standard credentials? There are a handful of sites where I have inconsistent access or have a double-NAT situation. I may be able to poke a server, but it may not be able to find its way back to the management system. Is there a workaround for this? The certificate configuration for HP SIM seems complicated. What is the preferred setup for trust between systems? I'd also appreciate any notes or recommendations to using this product. Or if there's a better way to do this, I'd like to know.

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  • How to get rid of a stubborn 'removed' device in mdadm

    - by T.J. Crowder
    One of my server's drives failed and so I removed the failed drive from all three relevant arrays, had the drive swapped out, and then added the new drive to the arrays. Two of the arrays worked perfectly. The third added the drive back as a spare, and there's an odd "removed" entry in the mdadm details. I tried both mdadm /dev/md2 --remove failed and mdadm /dev/md2 --remove detached as suggested here and here, neither of which complained, but neither of which had any effect, either. Does anyone know how I can get rid of that entry and get the drive added back properly? (Ideally without resyncing a third time, I've already had to do it twice and it takes hours. But if that's what it takes, that's what it takes.) The new drive is /dev/sda, the relevant partition is /dev/sda3. Here's the detail on the array: # mdadm --detail /dev/md2 /dev/md2: Version : 0.90 Creation Time : Wed Oct 26 12:27:49 2011 Raid Level : raid1 Array Size : 729952192 (696.14 GiB 747.47 GB) Used Dev Size : 729952192 (696.14 GiB 747.47 GB) Raid Devices : 2 Total Devices : 2 Preferred Minor : 2 Persistence : Superblock is persistent Update Time : Tue Nov 12 17:48:53 2013 State : clean, degraded Active Devices : 1 Working Devices : 2 Failed Devices : 0 Spare Devices : 1 UUID : 2fdbf68c:d572d905:776c2c25:004bd7b2 (local to host blah) Events : 0.34665 Number Major Minor RaidDevice State 0 0 0 0 removed 1 8 19 1 active sync /dev/sdb3 2 8 3 - spare /dev/sda3 If it's relevant, it's a 64-bit server. It normally runs Ubuntu, but right now I'm in the data centre's "rescue" OS, which is Debian 7 (wheezy). The "removed" entry was there the last time I was in Ubuntu (it won't, currently, boot from the disk), so I don't think that's not some Ubuntu/Debian conflict (and they are, of course, closely related). Update: Having done extensive tests with test devices on a local machine, I'm just plain getting anomalous behavior from mdadm with this array. For instance, with /dev/sda3 removed from the array again, I did this: mdadm /dev/md2 --grow --force --raid-devices=1 And that got rid of the "removed" device, leaving me just with /dev/sdb3. Then I nuked /dev/sda3 (wrote a file system to it, so it didn't have the raid fs anymore), then: mdadm /dev/md2 --grow --raid-devices=2 ...which gave me an array with /dev/sdb3 in slot 0 and "removed" in slot 1 as you'd expect. Then mdadm /dev/md2 --add /dev/sda3 ...added it — as a spare again. (Another 3.5 hours down the drain.) So with the rebuilt spare in the array, given that mdadm's man page says RAID-DEVICES CHANGES ... When the number of devices is increased, any hot spares that are present will be activated immediately. ...I grew the array to three devices, to try to activate the "spare": mdadm /dev/md2 --grow --raid-devices=3 What did I get? Two "removed" devices, and the spare. And yet when I do this with a test array, I don't get this behavior. So I nuked /dev/sda3 again, used it to create a brand-new array, and am copying the data from the old array to the new one: rsync -r -t -v --exclude 'lost+found' --progress /mnt/oldarray/* /mnt/newarray This will, of course, take hours. Hopefully when I'm done, I can stop the old array entirely, nuke /dev/sdb3, and add it to the new array. Hopefully, it won't get added as a spare!

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