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  • Managing Operational Risk of Financial Services Processes – part 2/2

    - by Sanjeev Sharma
    Normal 0 false false false EN-US X-NONE X-NONE MicrosoftInternetExplorer4 /* Style Definitions */ table.MsoNormalTable {mso-style-name:"Table Normal"; mso-tstyle-rowband-size:0; mso-tstyle-colband-size:0; mso-style-noshow:yes; mso-style-priority:99; mso-style-qformat:yes; mso-style-parent:""; mso-padding-alt:0in 5.4pt 0in 5.4pt; mso-para-margin:0in; mso-para-margin-bottom:.0001pt; mso-pagination:widow-orphan; font-size:10.0pt; font-family:"Calibri","sans-serif"; mso-bidi-font-family:"Times New Roman";} In my earlier blog post, I had described the factors that lead to compliance complexity of financial services processes. In this post, I will outline the business implications of the increasing process compliance complexity and the specific role of BPM in addressing the operational risk reduction objectives of regulatory compliance. First, let’s look at the business implications of increasing complexity of process compliance for financial institutions: · Increased time and cost of compliance due to duplication of effort in conforming to regulatory requirements due to process changes driven by evolving regulatory mandates, shifting business priorities or internal/external audit requirements · Delays in audit reporting due to quality issues in reconciling non-standard process KPIs and integrity concerns arising from the need to rely on multiple data sources for a given process Next, let’s consider some approaches to managing the operational risk of business processes. Financial institutions considering reducing operational risk of their processes, generally speaking, have two choices: · Rip-and-replace existing applications with new off-the shelf applications. · Extend capabilities of existing applications by modeling their data and process interactions, with other applications or user-channels, outside of the application boundary using BPM. The benefit of the first approach is that compliance with new regulatory requirements would be embedded within the boundaries of these applications. However pre-built compliance of any packaged application or custom-built application should not be mistaken as a one-shot fix for future compliance needs. The reason is that business needs and regulatory requirements inevitably out grow end-to-end capabilities of even the most comprehensive packaged or custom-built business application. Thus, processes that originally resided within the application will eventually spill outside the application boundary. It is precisely at such hand-offs between applications or between overlaying processes where vulnerabilities arise to unknown and accidental faults that potentially result in errors and lead to partial or total failure. The gist of the above argument is that processes which reside outside application boundaries, in other words, span multiple applications constitute a latent operational risk that spans the end-to-end value chain. For instance, distortion of data flowing from an account-opening application to a credit-rating system if left un-checked renders compliance with “KYC” policies void even when the “KYC” checklist was enforced at the time of data capture by the account-opening application. Oracle Business Process Management is enabling financial institutions to lower operational risk of such process ”gaps” for Financial Services processes including “Customer On-boarding”, “Quote-to-Contract”, “Deposit/Loan Origination”, “Trade Exceptions”, “Interest Claim Tracking” etc.. If you are faced with a similar challenge and need any guidance on the same feel free to drop me a note.

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  • We have our standards, and we need them

    - by Tony Davis
    The presenter suddenly broke off. He was midway through his section on how to apply to the relational database the Continuous Delivery techniques that allowed for rapid-fire rounds of development and refactoring, while always retaining a “production-ready” state. He sighed deeply and then launched into an astonishing diatribe against Database Administrators, much of his frustration directed toward Oracle DBAs, in particular. In broad strokes, he painted the picture of a brave new deployment philosophy being frustratingly shackled by the relational database, and by especially by the attitudes of the guardians of these databases. DBAs, he said, shunned change and “still favored tools I’d have been embarrassed to use in the ’80′s“. DBAs, Oracle DBAs especially, were more attached to their vendor than to their employer, since the former was the primary source of their career longevity and spectacular remuneration. He contended that someone could produce the best IDE or tool in the world for Oracle DBAs and yet none of them would give a stuff, unless it happened to come from the “mother ship”. I sat blinking in astonishment at the speaker’s vehemence, and glanced around nervously. Nobody in the audience disagreed, and a few nodded in assent. Although the primary target of the outburst was the Oracle DBA, it made me wonder. Are we who work with SQL Server, database professionals or merely SQL Server fanbois? Do DBAs, in general, have an image problem? Is it a good career-move to be seen to be holding onto a particular product by the whites of our knuckles, to the exclusion of all else? If we seek a broad, open-minded, knowledge of our chosen technology, the database, and are blessed with merely mortal powers of learning, then we like standards. Vendors of RDBMSs generally don’t conform to standards by instinct, but by customer demand. Microsoft has made great strides to adopt the international SQL Standards, where possible, thanks to considerable lobbying by the community. The implementation of Window functions is a great example. There is still work to do, though. SQL Server, for example, has an unusable version of the Information Schema. One cast-iron rule of any RDBMS is that we must be able to query the metadata using the same language that we use to query the data, i.e. SQL, and we do this by running queries against the INFORMATION_SCHEMA views. Developers who’ve attempted to apply a standard query that works on MySQL, or some other database, but doesn’t produce the expected results on SQL Server are advised to shun the Standards-based approach in favor of the vendor-specific one, using the catalog views. The argument behind this is sound and well-documented, and of course we all use those catalog views, out of necessity. And yet, as database professionals, committed to supporting the best databases for the business, whatever they are now and in the future, surely our heart should sink somewhat when we advocate a vendor specific approach, to a developer struggling with something as simple as writing a guard clause. And when we read messages on the Microsoft documentation informing us that we shouldn’t rely on INFORMATION_SCHEMA to identify reliably the schema of an object, in SQL Server!

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  • ORE graphics using Remote Desktop Protocol

    - by Sherry LaMonica
    Oracle R Enterprise graphics are returned as raster, or bitmap graphics. Raster images consist of tiny squares of color information referred to as pixels that form points of color to create a complete image. Plots that contain raster images render quickly in R and create small, high-quality exported image files in a wide variety of formats. However, it is a known issue that the rendering of raster images can be problematic when creating graphics using a Remote Desktop connection. Raster images do not display in the windows device using Remote Desktop under the default settings. This happens because Remote Desktop restricts the number of colors when connecting to a Windows machine to 16 bits per pixel, and interpolating raster graphics requires many colors, at least 32 bits per pixel.. For example, this simple embedded R image plot will be returned in a raster-based format using a standalone Windows machine:  R> library(ORE) R> ore.connect(user="rquser", sid="orcl", host="localhost", password="rquser", all=TRUE)  R> ore.doEval(function() image(volcano, col=terrain.colors(30))) Here, we first load the ORE packages and connect to the database instance using database login credentials. The ore.doEval function executes the R code within the database embedded R engine and returns the image back to the client R session. Over a Remote Desktop connection under the default settings, this graph will appear blank due to the restricted number of colors. Users who encounter this issue have two options to display ORE graphics over Remote Desktop: either raise Remote Desktop's Color Depth or direct the plot output to an alternate device. Option #1: Raise Remote Desktop Color Depth setting In a Remote Desktop session, all environment variables, including display variables determining Color Depth, are determined by the RCP-Tcp connection settings. For example, users can reduce the Color Depth when connecting over a slow connection. The different settings are 15 bits, 16 bits, 24 bits, or 32 bits per pixel. To raise the Remote Desktop color depth: On the Windows server, launch Remote Desktop Session Host Configuration from the Accessories menu.Under Connections, right click on RDP-Tcp and select Properties.On the Client Settings tab either uncheck LimitMaximum Color Depth or set it to 32 bits per pixel. Click Apply, then OK, log out of the remote session and reconnect.After reconnecting, the Color Depth on the Display tab will be set to 32 bits per pixel.  Raster graphics will now display as expected. For ORE users, the increased color depth results in slightly reduced performance during plot creation, but the graph will be created instead of displaying an empty plot. Option #2: Direct plot output to alternate device Plotting to a non-windows device is a good option if it's not possible to increase Remote Desktop Color Depth, or if performance is degraded when creating the graph. Several device drivers are available for off-screen graphics in R, such as postscript, pdf, and png. On-screen devices include windows, X11 and Cairo. Here we output to the Cairo device to render an on-screen raster graphic.  The grid.raster function in the grid package is analogous to other grid graphical primitives - it draws a raster image within the current plot's grid.  R> options(device = "CairoWin") # use Cairo device for plotting during the session R> library(Cairo) # load Cairo, grid and png libraries  R> library(grid) R> library(png)  R> res <- ore.doEval(function()image(volcano,col=terrain.colors(30))) # create embedded R plot  R> img <- ore.pull(res, graphics = TRUE)$img[[1]] # extract image  R> grid.raster(as.raster(readPNG(img)), interpolate = FALSE) # generate raster graph R> dev.off() # turn off first device   By default, the interpolate argument to grid.raster is TRUE, which means that what is actually drawn by R is a linear interpolation of the pixels in the original image. Setting interpolate to FALSE uses a sample from the pixels in the original image.A list of graphics devices available in R can be found in the Devices help file from the grDevices package: R> help(Devices)

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  • Missing Fields and Default Values

    - by PointsToShare
    © 2011 By: Dov Trietsch. All rights reserved Dealing with Missing Fields and Default Values New fields and new default values are not propagated throughout the list. They only apply to new and updated items and not to items already entered. They are only prospective. We need to be able to deal with this issue. Here is a scenario. The user has an old list with old items and adds a new field. The field is not created for any of the old items. Trying to get its value raises an Argument Exception. Here is another: a default value is added to a field. All the old items, where the field was not assigned a value, do not get the new default value. The two can also happen in tandem – a new field is added with a default. The older items have neither. Even better, if the user changes the default value, the old items still carry the old defaults. Let’s go a bit further. You have already written code for the list, be it an event receiver, a feature receiver, a console app or a command extension, in which you span all the fields and run on selected items – some new (no problem) and some old (problems aplenty). Had you written defensive code, you would be able to handle the situation, including similar changes in the future. So, without further ado, here’s how. Instead of just getting the value of a field in an item – item[field].ToString() – use the function below. I use ItemValue(item, fieldname, “mud in your eye”) and if “mud in your eye” is what I get, I know that the item did not have the field.   /// <summary> /// Return the column value or a default value /// </summary> private static string ItemValue(SPItem item, string column, string defaultValue) {     try     {         return item[column].ToString();     }     catch (NullReferenceException ex)     {         return defaultValue;     }     catch (ArgumentException ex)     {         return defaultValue;     } } I also use a similar function to return the default and a funny default-default to ascertain that the default does not exist. Here it is:  /// <summary> /// return a fields default or the "default" default. /// </summary> public static string GetFieldDefault(SPField fld, string defValue) {     try     {         // -- Check if default exists.         return fld.DefaultValue.ToString();     }     catch (NullReferenceException ex)     {         return defValue;     }     catch (ArgumentException ex)     {         return defValue;     } } How is this defensive? You have trapped an expected error and dealt with it. Therefore the program did not stop cold in its track and the required code ran to its end. Now, take a further step - write to a log (See Logging – a log blog). Read your own log every now and then, and act accordingly. That’s all Folks!

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  • Move over DFS and Robocopy, here is SyncToy!

    - by andywe
    Ever since Windows 2000, I have always had the need to replicate data to multiple endpoints with the same content. Until DFS was introduced, the method of thinking was to either manually copy the data location by location, or to batch script it with xcopy and schedule a task. Even though this worked (and still does today), it was cumbersome, and intensive on the network, especially when dealing with larger amounts of data. Then along came robocopy, as an internal tool written by an enterprising programmer at Microsoft. We used it quite a bit, especially when we could not use DFS in the early days. It was received so well, it made it into the public realm. At least now we could have the ability to determine what files had changed and only replicate those. Well, over time there has been evolution of this ideal. DFS is obviously the Windows enterprise class service to do this, along with BrancheCache..however you don’t always need or want the power of DFS, especially when it comes to small datacenter installations, or remote offices. I have specific data sets that are on closed or restricted networks, that either have a security need for this, or are in remote countries where bandwidth is a premium. FOr this, I use the latest evolution for one off replication names Synctoy. Synctoy is from Microsoft, seemingly released in 2009, that wraps a nice GUI around setting up a paired set of folders (remember the mobile briefcase from Windows 98?), and allowing you the choice of synchronization methods. 1 way, or 2 way. Simply create a paired set of folders on the source and destination, choose your options for content, exclude any file types you don’t want to replicate, and click run. Scheduling is even easier. MS has included a wrapper for doing just this so all you enter in your task schedule in the SynToyCMD.exe, a –R as an argument, and the time schedule. No more complicated command lines or scripts.   I find this especially useful when I use MS backup to back up a system volume, but only want subsets of backup information of a data share and ONLY when that dataset has changed. Not relying on full backups and incremental. An example of this is my application installation master share. I back this up with SyncToy because I do not need multiple backup copies..one copy elsewhere suffices to back it up. At home, very useful for your pictures, videos, music, ect..the backup is online and ready to access, not waiting for you to restore a backup file, and no need to institute a domain simply to have DFS.'   Do note there is a risk..if you accidently delete a file and do not catch this before the next sync, then depending on your SyncToy settings, you can indeed lose that file as the destination updates..so due diligence applies. I make it a rule to sync manly one way…I use my master share for making changes, and allow the schedule to follow suit. Any real important file I lock down as read only through file permissions so it cannot be deleted unless I intervene.   Check out the tool and have some fun! http://www.microsoft.com/en-us/download/details.aspx?DisplayLang=en&id=15155

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  • Some PowerShell goodness

    - by KyleBurns
    Ever work somewhere where processes dump files into folders to maintain an archive?  Me too and Windows Explorer hates it.  Very often I find myself needing to organize these files into subfolders so that I can go after files without locking up Windows Explorer and my answer used to be to write a program in something like C# to do the job.  These programs will typically enumerate the files in a folder and move each file to a subdirectory named based on a datestamp.  The last such program I wrote had to use lower-level Win32 API calls to perform the enumeration because it appears the standard .Net calls make use of the same method of enumerating the directories that Windows Explorer chokes on when dealing with a large number of entries in a particular directory, so a simple task was accomplished with a lot of code. Of course, this little utility was just something I used to make my life easier and "not a production app", so it was in my local source folder and not source control when my hard drive died.  So... I was getting ready to re-create it and thought it might be a good idea to play with PowerShell a bit - something I had been wanting to do but had not yet met a requirement to make me do it.  The resulting script was amazingly succinct and even building the flexibility for parameterization and adding line breaks for readability was only about 25 lines long.  Here's the code with discussion following: param(     [Parameter(         Mandatory = $false,         Position = 0,         HelpMessage = "Root of the folders or share to archive.  Be sure to end with appropriate path separator"     )]     [String] $folderRoot="\\fileServer\pathToFolderWithLotsOfFiles\",       [Parameter(         Mandatory = $false,         Position = 1     )]     [int] $days = 1 ) dir $folderRoot|?{(!($_.PsIsContainer)) -and ((get-date) - $_.lastwritetime).totaldays -gt $days }|%{     [string]$year=$([string]$_.lastwritetime.year)     [string]$month=$_.lastwritetime.month     [string]$day=$_.lastwritetime.day     $dir=$folderRoot+$year+"\"+$month+"\"+$day     if(!(test-path $dir)){         new-item -type container $dir     }     Write-output $_     move-item $_.fullname $dir } The script starts by declaring two parameters.  The first parameter holds the path to the folder that I am going to be sorting into subdirectories.  The path separator is intended to be included in this argument because I didn't want to mess with determining whether this was local or UNC and picking the right separator in code, but this could be easily improved upon using Path.Combine since PowerShell has access to the full framework libraries.  The second parameter holds a minimum age in days for files to be removed from the root folder.  The script then pipes the dir command through a query to include only files (by excluding containers) and of those, only entries that meet the age requirement based on the last modified datestamp.  For each of those, the datestamp is used to construct a folder name in the format YYYY\MM\DD (if you're in an environment where even a day's worth of files need further divided, you could make this more granular) and the folder is created if it does not yet exist.  Finally, the file is moved into the directory. One of the things that was really cool about using PowerShell for this task is that the new-item command is smart enough to create the entire subdirectory structure with a single call.  In previous code that I have written to do this kind of thing, I would have to test the entire tree leading down to the subfolder I want, leading to a lot of branching code that detracted from being able to quickly look at the code and understand the job it performs. Overall, I have to say I'm really pleased with what has been done making PowerShell powerful and useful.

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  • Planning development when academic research is involved

    - by Another Anonymous User
    Dear fellow programmers, how do you do "software planning" when academic research is involved? And, on a side note, how do you convince your boss that writing software is not like building a house and it's more like writing a novel? The gory details are below. I am in charge of a small dev team working in a research lab. We started developing a software with the purpose of going public one day (i.e. sell and make money off that). Such software depends on, amongst other things, at least two independent research lines: that is, there are at least two Ph.D. candidates that will, hopefully, one day come out with a working implementation of what we need. The main software depends also on other, more concrete resources that we as developers can take care of: graphics rendering, soft bodies deformation, etc. My boss asked me to write the specifications, requirements AND a bloody GANTT chart of the entire project. Faced with the fact that I don't have a clue about the research part, and that such research is fundamental for the software, he said "make assumptions." For the clarity of the argument, he is a professor whose Ph.D. students should come up with the research we need. And he comes from a strictly engineering background: plan everything first, write down specifications and only then write down code that "it's the last part". What I am doing now: I broke down the product in features; each 'feature' is, de facto, a separate product; Each feature is built on top of the previous one; Once a feature (A) has a working prototype the team can start working on the next feature (B), while QA for is being done for A (if money allows, more people can be brought in, etc.); Features that depend on research will come last: by then, hopefully, the research part will be completed (when is still a big question) ; Also, I set the team to use SCRUM for the development of 'version 1.0', due in a few months. This deadline could be set based on reasonable assumptions: we listed all required features, we counted our availability, and we gave a reasonable estimate. So my questions, again, are: How do I make my boss happy while at the same time get something out the door? How do I write specifications for something we -the developers- have no clue whether it's possible to do or not? (We still haven't decided which libraries to use for some tasks; we'll do so when we'll need to) How do I get the requirements for that, given that there are yet no clients nor investors, just lots of interests and promises? How do I get peace in the world? I am sure at least one of my questions will be answered :) ps: I am writing this anonymously since a potential investor might backfire if this is discovered. Hope you'll understand. However I must say I do not like this mentality of 'hiding the truth': this program will likely benefit many, and not being able to talk openly about this (with my name and my reputation attached) feels like censorship. But alas, I care more about your suggestions now.

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  • Data breakpoints to find points where data gets broken

    - by raccoon_tim
    When working with a large code base, finding reasons for bizarre bugs can often be like finding a needle in a hay stack. Finding out why an object gets corrupted without no apparent reason can be quite daunting, especially when it seems to happen randomly and totally out of context. Scenario Take the following scenario as an example. You have defined the a class that contains an array of characters that is 256 characters long. You now implement a method for filling this buffer with a string passed as an argument. At this point you mistakenly expect the buffer to be 256 characters long. At some point you notice that you require another character buffer and you add that after the previous one in the class definition. You now figure that you don’t need the 256 characters that the first member can hold and you shorten that to 128 to conserve space. At this point you should start thinking that you also have to modify the method defined above to safeguard against buffer overflow. It so happens, however, that in this not so perfect world this does not cross your mind. Buffer overflow is one of the most frequent sources for errors in a piece of software and often one of the most difficult ones to detect, especially when data is read from an outside source. Many mass copy functions provided by the C run-time provide versions that have boundary checking (defined with the _s suffix) but they can not guard against hard coded buffer lengths that at some point get changed. Finding the bug Getting back to the scenario, you’re now wondering why does the second string get modified with data that makes no sense at all. Luckily, Visual Studio provides you with a tool to help you with finding just these kinds of errors. It’s called data breakpoints. To add a data breakpoint, you first run your application in debug mode or attach to it in the usual way, and then go to Debug, select New Breakpoint and New Data Breakpoint. In the popup that opens, you can type in the memory address and the amount of bytes you wish to monitor. You can also use an expression here, but it’s often difficult to come up with an expression for data in an object allocated on the heap when not in the context of a certain stack frame. There are a couple of things to note about data breakpoints, however. First of all, Visual Studio supports a maximum of four data breakpoints at any given time. Another important thing to notice is that some C run-time functions modify memory in kernel space which does not trigger the data breakpoint. For instance, calling ReadFile on a buffer that is monitored by a data breakpoint will not trigger the breakpoint. The application will now break at the address you specified it to. Often you might immediately spot the issue but the very least this feature can do is point you in the right direction in search for the real reason why the memory gets inadvertently modified. Conclusions Data breakpoints are a great feature, especially when doing a lot of low level operations where multiple locations modify the same data. With the exception of some special cases, like kernel memory modification, you can use it whenever you need to check when memory at a certain location gets changed on purpose or inadvertently.

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  • Why is x=x++ undefined?

    - by ugoren
    It's undefined because the it modifies x twice between sequence points. The standard says it's undefined, therefore it's undefined. That much I know. But why? My understanding is that forbidding this allows compilers to optimize better. This could have made sense when C was invented, but now seems like a weak argument. If we were to reinvent C today, would we do it this way, or can it be done better? Or maybe there's a deeper problem, that makes it hard to define consistent rules for such expressions, so it's best to forbid them? So suppose we were to reinvent C today. I'd like to suggest simple rules for expressions such as x=x++, which seem to me to work better than the existing rules. I'd like to get your opinion on the suggested rules compared to the existing ones, or other suggestions. Suggested Rules: Between sequence points, order of evaluation is unspecified. Side effects take place immediately. There's no undefined behavior involved. Expressions evaluate to this value or that, but surely won't format your hard disk (strangely, I've never seen an implementation where x=x++ formats the hard disk). Example Expressions x=x++ - Well defined, doesn't change x. First, x is incremented (immediately when x++ is evaluated), then it's old value is stored in x. x++ + ++x - Increments x twice, evaluates to 2*x+2. Though either side may be evaluated first, the result is either x + (x+2) (left side first) or (x+1) + (x+1) (right side first). x = x + (x=3) - Unspecified, x set to either x+3 or 6. If the right side is evaluated first, it's x+3. It's also possible that x=3 is evaluated first, so it's 3+3. In either case, the x=3 assignment happens immediately when x=3 is evaluated, so the value stored is overwritten by the other assignment. x+=(x=3) - Well defined, sets x to 6. You could argue that this is just shorthand for the expression above. But I'd say that += must be executed after x=3, and not in two parts (read x, evaluate x=3, add and store new value). What's the Advantage? Some comments raised this good point. It's not that I'm after the pleasure of using x=x++ in my code. It's a strange and misleading expression. What I want is to be able to understand complicated expressions. Normally, a complicated expression is no more than the sum of its parts. If you understand the parts and the operators combining them, you can understand the whole. C's current behavior seems to deviate from this principle. One assignment plus another assignment suddenly doesn't make two assignments. Today, when I look at x=x++, I can't say what it does. With my suggested rules, I can, by simply examining its components and their relations.

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  • Is this over-abstraction? (And is there a name for it?)

    - by mwhite
    I work on a large Django application that uses CouchDB as a database and couchdbkit for mapping CouchDB documents to objects in Python, similar to Django's default ORM. It has dozens of model classes and a hundred or two CouchDB views. The application allows users to register a "domain", which gives them a unique URL containing the domain name that gives them access to a project whose data has no overlap with the data of other domains. Each document that is part of a domain has its domain property set to that domain's name. As far as relationships between the documents go, all domains are effectively mutually exclusive subsets of the data, except for a few edge cases (some users can be members of more than one domain, and there are some administrative reports that include all domains, etc.). The code is full of explicit references to the domain name, and I'm wondering if it would be worth the added complexity to abstract this out. I'd also like to know if there's a name for the sort of bound property approach I'm taking here. Basically, I have something like this in mind: Before in models.py class User(Document): domain = StringProperty() class Group(Document): domain = StringProperty() name = StringProperty() user_ids = StringListProperty() # method that returns related document set def users(self): return [User.get(id) for id in self.user_ids] # method that queries a couch view optimized for a specific lookup @classmethod def by_name(cls, domain, name): # the view method is provided by couchdbkit and handles # wrapping json CouchDB results as Python objects, and # can take various parameters modifying behavior return cls.view('groups/by_name', key=[domain, name]) # method that creates a related document def get_new_user(self): user = User(domain=self.domain) user.save() self.user_ids.append(user._id) return user in views.py: from models import User, Group # there are tons of views like this, (request, domain, ...) def create_new_user_in_group(request, domain, group_name): group = Group.by_name(domain, group_name)[0] user = User(domain=domain) user.save() group.user_ids.append(user._id) group.save() in group/by_name/map.js: function (doc) { if (doc.doc_type == "Group") { emit([doc.domain, doc.name], null); } } After models.py class DomainDocument(Document): domain = StringProperty() @classmethod def domain_view(cls, *args, **kwargs): kwargs['key'] = [cls.domain.default] + kwargs['key'] return super(DomainDocument, cls).view(*args, **kwargs) @classmethod def get(cls, *args, **kwargs, validate_domain=True): ret = super(DomainDocument, cls).get(*args, **kwargs) if validate_domain and ret.domain != cls.domain.default: raise Exception() return ret def models(self): # a mapping of all models in the application. accessing one returns the equivalent of class BoundUser(User): domain = StringProperty(default=self.domain) class User(DomainDocument): pass class Group(DomainDocument): name = StringProperty() user_ids = StringListProperty() def users(self): return [self.models.User.get(id) for id in self.user_ids] @classmethod def by_name(cls, name): return cls.domain_view('groups/by_name', key=[name]) def get_new_user(self): user = self.models.User() user.save() views.py @domain_view # decorator that sets request.models to the same sort of object that is returned by DomainDocument.models and removes the domain argument from the URL router def create_new_user_in_group(request, group_name): group = request.models.Group.by_name(group_name) user = request.models.User() user.save() group.user_ids.append(user._id) group.save() (Might be better to leave the abstraction leaky here in order to avoid having to deal with a couchapp-style //! include of a wrapper for emit that prepends doc.domain to the key or some other similar solution.) function (doc) { if (doc.doc_type == "Group") { emit([doc.name], null); } } Pros and Cons So what are the pros and cons of this? Pros: DRYer prevents you from creating related documents but forgetting to set the domain. prevents you from accidentally writing a django view - couch view execution path that leads to a security breach doesn't prevent you from accessing underlying self.domain and normal Document.view() method potentially gets rid of the need for a lot of sanity checks verifying whether two documents whose domains we expect to be equal are. Cons: adds some complexity hides what's really happening requires no model modules to have classes with the same name, or you would need to add sub-attributes to self.models for modules. However, requiring project-wide unique class names for models should actually be fine because they correspond to the doc_type property couchdbkit uses to decide which class to instantiate them as, which should be unique. removes explicit dependency documentation (from group.models import Group)

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  • Is the Leptonica implementation of 'Modified Median Cut' not using the median at all?

    - by TheCodeJunkie
    I'm playing around a bit with image processing and decided to read up on how color quantization worked and after a bit of reading I found the Modified Median Cut Quantization algorithm. I've been reading the code of the C implementation in Leptonica library and came across something I thought was a bit odd. Now I want to stress that I am far from an expert in this area, not am I a math-head, so I am predicting that this all comes down to me not understanding all of it and not that the implementation of the algorithm is wrong at all. The algorithm states that the vbox should be split along the lagest axis and that it should be split using the following logic The largest axis is divided by locating the bin with the median pixel (by population), selecting the longer side, and dividing in the center of that side. We could have simply put the bin with the median pixel in the shorter side, but in the early stages of subdivision, this tends to put low density clusters (that are not considered in the subdivision) in the same vbox as part of a high density cluster that will outvote it in median vbox color, even with future median-based subdivisions. The algorithm used here is particularly important in early subdivisions, and 3is useful for giving visible but low population color clusters their own vbox. This has little effect on the subdivision of high density clusters, which ultimately will have roughly equal population in their vboxes. For the sake of the argument, let's assume that we have a vbox that we are in the process of splitting and that the red axis is the largest. In the Leptonica algorithm, on line 01297, the code appears to do the following Iterate over all the possible green and blue variations of the red color For each iteration it adds to the total number of pixels (population) it's found along the red axis For each red color it sum up the population of the current red and the previous ones, thus storing an accumulated value, for each red note: when I say 'red' I mean each point along the axis that is covered by the iteration, the actual color may not be red but contains a certain amount of red So for the sake of illustration, assume we have 9 "bins" along the red axis and that they have the following populations 4 8 20 16 1 9 12 8 8 After the iteration of all red bins, the partialsum array will contain the following count for the bins mentioned above 4 12 32 48 49 58 70 78 86 And total would have a value of 86 Once that's done it's time to perform the actual median cut and for the red axis this is performed on line 01346 It iterates over bins and check they accumulated sum. And here's the part that throws me of from the description of the algorithm. It looks for the first bin that has a value that is greater than total/2 Wouldn't total/2 mean that it is looking for a bin that has a value that is greater than the average value and not the median ? The median for the above bins would be 49 The use of 43 or 49 could potentially have a huge impact on how the boxes are split, even though the algorithm then proceeds by moving to the center of the larger side of where the matched value was.. Another thing that puzzles me a bit is that the paper specified that the bin with the median value should be located, but does not mention how to proceed if there are an even number of bins.. the median would be the result of (a+b)/2 and it's not guaranteed that any of the bins contains that population count. So this is what makes me thing that there are some approximations going on that are negligible because of how the split actually takes part at the center of the larger side of the selected bin. Sorry if it got a bit long winded, but I wanted to be as thoroughas I could because it's been driving me nuts for a couple of days now ;)

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  • High Resolution Timeouts

    - by user12607257
    The default resolution of application timers and timeouts is now 1 msec in Solaris 11.1, down from 10 msec in previous releases. This improves out-of-the-box performance of polling and event based applications, such as ticker applications, and even the Oracle rdbms log writer. More on that in a moment. As a simple example, the poll() system call takes a timeout argument in units of msec: System Calls poll(2) NAME poll - input/output multiplexing SYNOPSIS int poll(struct pollfd fds[], nfds_t nfds, int timeout); In Solaris 11, a call to poll(NULL,0,1) returns in 10 msec, because even though a 1 msec interval is requested, the implementation rounds to the system clock resolution of 10 msec. In Solaris 11.1, this call returns in 1 msec. In specification lawyer terms, the resolution of CLOCK_REALTIME, introduced by POSIX.1b real time extensions, is now 1 msec. The function clock_getres(CLOCK_REALTIME,&res) returns 1 msec, and any library calls whose man page explicitly mention CLOCK_REALTIME, such as nanosleep(), are subject to the new resolution. Additionally, many legacy functions that pre-date POSIX.1b and do not explicitly mention a clock domain, such as poll(), are subject to the new resolution. Here is a fairly comprehensive list: nanosleep pthread_mutex_timedlock pthread_mutex_reltimedlock_np pthread_rwlock_timedrdlock pthread_rwlock_reltimedrdlock_np pthread_rwlock_timedwrlock pthread_rwlock_reltimedwrlock_np mq_timedreceive mq_reltimedreceive_np mq_timedsend mq_reltimedsend_np sem_timedwait sem_reltimedwait_np poll select pselect _lwp_cond_timedwait _lwp_cond_reltimedwait semtimedop sigtimedwait aiowait aio_waitn aio_suspend port_get port_getn cond_timedwait cond_reltimedwait setitimer (ITIMER_REAL) misc rpc calls, misc ldap calls This change in resolution was made feasible because we made the implementation of timeouts more efficient a few years back when we re-architected the callout subsystem of Solaris. Previously, timeouts were tested and expired by the kernel's clock thread which ran 100 times per second, yielding a resolution of 10 msec. This did not scale, as timeouts could be posted by every CPU, but were expired by only a single thread. The resolution could be changed by setting hires_tick=1 in /etc/system, but this caused the clock thread to run at 1000 Hz, which made the potential scalability problem worse. Given enough CPUs posting enough timeouts, the clock thread could be a performance bottleneck. We fixed that by re-implementing the timeout as a per-CPU timer interrupt (using the cyclic subsystem, for those familiar with Solaris internals). This decoupled the clock thread frequency from timeout resolution, and allowed us to improve default timeout resolution without adding CPU overhead in the clock thread. Here are some exceptions for which the default resolution is still 10 msec. The thread scheduler's time quantum is 10 msec by default, because preemption is driven by the clock thread (plus helper threads for scalability). See for example dispadmin, priocntl, fx_dptbl, rt_dptbl, and ts_dptbl. This may be changed using hires_tick. The resolution of the clock_t data type, primarily used in DDI functions, is 10 msec. It may be changed using hires_tick. These functions are only used by developers writing kernel modules. A few functions that pre-date POSIX CLOCK_REALTIME mention _SC_CLK_TCK, CLK_TCK, "system clock", or no clock domain. These functions are still driven by the clock thread, and their resolution is 10 msec. They include alarm, pcsample, times, clock, and setitimer for ITIMER_VIRTUAL and ITIMER_PROF. Their resolution may be changed using hires_tick. Now back to the database. How does this help the Oracle log writer? Foreground processes post a redo record to the log writer, which releases them after the redo has committed. When a large number of foregrounds are waiting, the release step can slow down the log writer, so under heavy load, the foregrounds switch to a mode where they poll for completion. This scales better because every foreground can poll independently, but at the cost of waiting the minimum polling interval. That was 10 msec, but is now 1 msec in Solaris 11.1, so the foregrounds process transactions faster under load. Pretty cool.

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  • Reading graph inputs for a programming puzzle and then solving it

    - by Vrashabh
    I just took a programming competition question and I absolutely bombed it. I had trouble right at the beginning itself from reading the input set. The question was basically a variant of this puzzle http://codercharts.com/puzzle/evacuation-plan but also had an hour component in the first line(say 3 hours after start of evacuation). It reads like this This puzzle is a tribute to all the people who suffered from the earthquake in Japan. The goal of this puzzle is, given a network of road and locations, to determine the maximum number of people that can be evacuated. The people must be evacuated from evacuation points to rescue points. The list of road and the number of people they can carry per hour is provided. Input Specifications Your program must accept one and only one command line argument: the input file. The input file is formatted as follows: the first line contains 4 integers n r s t n is the number of locations (each location is given by a number from 0 to n-1) r is the number of roads s is the number of locations to be evacuated from (evacuation points) t is the number of locations where people must be evacuated to (rescue points) the second line contains s integers giving the locations of the evacuation points the third line contains t integers giving the locations of the rescue points the r following lines contain to the road definitions. Each road is defined by 3 integers l1 l2 width where l1 and l2 are the locations connected by the road (roads are one-way) and width is the number of people per hour that can fit on the road Now look at the sample input set 5 5 1 2 3 0 3 4 0 1 10 0 2 5 1 2 4 1 3 5 2 4 10 The 3 in the first line is the additional component and is defined as the number of hours since the resuce has started which is 3 in this case. Now my solution was to use Dijisktras algorithm to find the shortest path between each of the rescue and evac nodes. Now my problem started with how to read the input set. I read the first line in python and stored the values in variables. But then I did not know how to store the values of the distance between the nodes and what DS to use and how to input it to say a standard implementation of dijikstras algorithm. So my question is two fold 1.) How do I take the input of such problems? - I have faced this problem in quite a few competitions recently and I hope I can get a simple code snippet or an explanation in java or python to read the data input set in such a way that I can input it as a graph to graph algorithms like dijikstra and floyd/warshall. Also a solution to the above problem would also help. 2.) How to solve this puzzle? My algorithm was: Find shortest path between evac points (in the above example it is 14 from 0 to 3) Multiply it by number of hours to get maximal number of saves Also the answer given for the variant for the input set was 24 which I dont understand. Can someone explain that also. UPDATE: I get how the answer is 14 in the given problem link - it seems to be just the shortest path between node 0 and 3. But with the 3 hour component how is the answer 24 UPDATE I get how it is 24 - its a complete graph traversal at every hour and this is how I solve it Hour 1 Node 0 to Node 1 - 10 people Node 0 to Node 2- 5 people TotalRescueCount=0 Node 1=10 Node 2= 5 Hour 2 Node 1 to Node 3 = 5(Rescued) Node 2 to Node 4 = 5(Rescued) Node 0 to Node 1 = 10 Node 0 to Node 2 = 5 Node 1 to Node 2 = 4 TotalRescueCount = 10 Node 1 = 10 Node 2= 5+4 = 9 Hour 3 Node 1 to Node 3 = 5(Rescued) Node 2 to Node 4 = 5+4 = 9(Rescued) TotalRescueCount = 9+5+10 = 24 It hard enough for this case , for multiple evac and rescue points how in the world would I write a pgm for this ?

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  • Managed c++ std::string not accessible in unmanaged c++

    - by Radhesham
    In unmanaged c++ dll i have a function which takes constant std::string as argument Prototype : void read ( const std::string &amp;imageSpec_ ) I call this function from managed c++ dll by passing a std::string. When i debug the unmanaged c++ code the parameter imageSpec_ shows the value correctly but does not allow me to copy that value in other variable. imageSpec_.copy( sFilename, 4052 ); It shows length of imageSpec_ as 0(zero). If i try copying like std::string sTempFileName(imageSpec_); this statement string new string is a empty string. But for std::string sTempFileName(imageSpec_.c_str()); this statement string gets copied correctly. i.e. with charpointer string is copied correctly. Copying this way will need a major change in unmanaged c++ code. I am building unmanaged code in Visual studio 6.0 and managed c++ in Visual studio 2008. Is there any specific setting or code change in managed c++ that will solve the issue?

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  • X509Certificate.CreateFromCertFile - the specified network password is not correct

    - by pcampbell
    I have a .NET application that I want to use as a client to call an SSL SOAP web service. I have been supplied with a valid client certificate called foo.pfx. There is a password on the certificate itself. I've located the certificate at the following location: C:\certs\foo.pfx To call the web service, I need to attach the client certificate. Here's the code: public X509Certificate GetCertificateFromDisk(){ try{ string certPath = ConfigurationManager.AppSettings["MyCertPath"].ToString(); //this evaluates to "c:\\certs\\foo.pfx". So far so good. X509Certificate myCert = X509Certificate.CreateFromCertFile(certPath); // exception is raised here! "The specified network password is not correct" return cert; } catch (Exception ex){ throw; } } It sounds like the exception is around the .NET application trying to read the disk. The method CreateFromCertFile is a static method that should create a new instance of X509Certificate. The method isn't overridden, and has only one argument: the path. When I inspect the Exception, I find this: _COMPlusExceptionCode = -532459699 Source=mscorlib Question: does anyone know what the cause of the exception "The specified network password is not correct" ?

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  • Use MTOM/streaming from C# calling a webservice in java exposed via jaxws

    - by raticulin
    We have this webservice created with jax-ws @WebService(name = "Mywebser", targetNamespace = "http://namespace") @MTOM(threshold = 2048) @SOAPBinding(style = SOAPBinding.Style.DOCUMENT, use = SOAPBinding.Use.LITERAL, parameterStyle = SOAPBinding.ParameterStyle.WRAPPED) public class Mywebser { @WebMethod(operationName = "doStreaming", action = "urn:doStreaming") @WebResult(name = "return") public ResultInfo doStreaming(String username, String pwd, @XmlMimeType("application/octet-stream") DataHandler data, boolean overw){ ... } } The generated client side looks like this: @WebMethod(action = "urn:doStreaming") @WebResult(targetNamespace = "") @RequestWrapper(localName = "doStreaming", targetNamespace = "http://namespace", className = "com.mypack.client.doStreaming") @ResponseWrapper(localName = "doStreamingResponse", targetNamespace = "http://namespace", className = "com.mypack.client.doStreamingResponse") public ResultInfo doStreaming( @WebParam(name = "arg0", targetNamespace = "") String arg0, @WebParam(name = "arg1", targetNamespace = "") String arg1, @WebParam(name = "arg2", targetNamespace = "") DataHandler arg2, @WebParam(name = "arg3", targetNamespace = "") boolean arg3); By using it this way it uses streaming properly (verified we can pass an argument of 80mb when the jvm had less allowed. MywebserService serv = ...; Mywebser wso = serv.getMywebserPort(new MTOMFeature()); Map<String, Object> ctxt = ((BindingProvider) wso).getRequestContext(); ctxt.put(JAXWSProperties.HTTP_CLIENT_STREAMING_CHUNK_SIZE, 8192); DataHandler dataHandler = new DataHandler(new FileDataSource("c:\\temp\\A.dat")); arcres = wso.doStreaming("a", "b", dataHandler, true); We generate a clienet for .net, with VS2008, using "Add Web Reference", we get this C# code: [System.Web.Services.Protocols.SoapDocumentMethodAttribute("urn:doStreaming",RequestNamespace="http://namespace",ResponseNamespace="http://namespace",Use=System.Web.Services.Description.SoapBindingUse.Literal,ParameterStyle=System.Web.Services.Protocols.SoapParameterStyle.Wrapped)] [return: System.Xml.Serialization.XmlElementAttribute("return",Form=System.Xml.Schema.XmlSchemaForm.Unqualified)] public ResultInfo doStreaming( [System.Xml.Serialization.XmlElementAttribute(Form=System.Xml.Schema.XmlSchemaForm.Unqualified)] string arg0, [System.Xml.Serialization.XmlElementAttribute(Form=System.Xml.Schema.XmlSchemaForm.Unqualified)] string arg1, [System.Xml.Serialization.XmlElementAttribute(Form=System.Xml.Schema.XmlSchemaForm.Unqualified,DataType="base64Binary")] byte[] arg2, [System.Xml.Serialization.XmlElementAttribute(Form=System.Xml.Schema.XmlSchemaForm.Unqualified)] bool arg3) Apparently this is not using streaming? The type base64Binary of arg2 seems not the right one? In java it's a DataHandler. By testing it with low memory on the java side we can see it is not using streaming as it fails with OOM. Does someone knows if this is possible, and if so how? Our environment: server: jdk1.6, jaxws 2.1.7 client: C# 2.0, visual studio 2008

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  • High Profile ASP.NET websites

    - by nandos
    About twice a month I get asked to justify the reason "Why are we using ASP.NET and not PHP or Java, or buzz-word-of-the-month-here, etc". 100% of the time the questions come from people that do not understand anything about technology. People that would not know the difference between FTP and HTTP. The best approach I found (so far) to justify it to people without getting into technical details is to just say "XXX website uses it". Which I get back "Oh...I did not know that, so ASP.NET must be good". I know, I know, it hurts. But it works. So, without getting into the merit of why I'm using ASP.NET (which could trigger an endless argument for other platforms), I'm trying to compile a list of high profile websites that are implemented in ASP.NET. (No, they would have no idea what StackOverflow is). Can you name a high-profile website implemented in ASP.NET? EDIT: Current list (thanks for all the responses): (trying to avoid tech sites and prioritizing retail sites) Costco - http://www.costco.com/ Crate & Barrel - http://www.crateandbarrel.com/ Home Shopping Network - http://www.hsn.com/ Buy.com - http://www.buy.com/ Dell - http://www.dell.com Nasdaq - http://www.nasdaq.com/ Virgin - http://www.virgin.com/ 7-Eleven - http://www.7-eleven.com/ Carnival Cruise Lines - http://www.carnival.com/ L'Oreal - http://www.loreal.com/ The White House - http://www.whitehouse.gov/ Remax - http://www.remax.com/ Monster Jobs - http://www.monster.com/ USA Today - http://www.usatoday.com/ ComputerJobs.com - http://computerjobs.com/ Match.com - http://www.match.com National Health Services (UK) - http://www.nhs.uk/ CarrerBuilder.com - http://www.careerbuilder.com/

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  • Spring 3.0 making JSON response using jackson message converter

    - by dupdup
    i configure my messageconverter as Jackson's then class Foo{int x; int y} and in controller @ResponseBody public Foo method(){ return new Foo(3,4) } from that i m expecting to return a JSON string {x:'3',y:'4'} from server without any other configuration. but getting 404 error response to my ajax request If the method is annotated with @ResponseBody, the return type is written to the response HTTP body. The return value will be converted to the declared method argument type using HttpMessageConverters. Am I wrong ? or should I convert my response Object to Json string myself using serializer and then returning that string as response.(I could make string responses correctly) or should I make some other configurations ? like adding annotations for class Foo here is my conf.xml <bean id="jacksonMessageConverter" class="org.springframework.http.converter.json.MappingJacksonHttpMessageConverter"> <bean class="org.springframework.web.servlet.mvc.annotation.AnnotationMethodHandlerAdapter"> <property name="messageConverters"> <list> <ref bean="jacksonMessageConverter"/> </list> </property>

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  • Python, unit test - Pass command line arguments to setUp of unittest.TestCase

    - by sberry2A
    I have a script that acts as a wrapper for some unit tests written using the Python unittest module. In addition to cleaning up some files, creating an output stream and generating some code, it loads test cases into a suite using unittest.TestLoader().loadTestsFromTestCase() I am already using optparse to pull out several command-line arguments used for determining the output location, whether to regenerate code and whether to do some clean up. I also want to pass a configuration variable, namely an endpoint URI, for use within the test cases. I realize I can add an OptionParser to the setUp method of the TestCase, but I want to instead pass the option to setUp. Is this possible using loadTestsFromTestCase()? I can iterate over the returned TestSuite's TestCases, but can I manually call setUp on the TestCases? ** EDIT ** I wanted to point out that I am able to pass the arguments to setUp if I iterate over the tests and call setUp manually like: (options, args) = op.parse_args() suite = unittest.TestLoader().loadTestsFromTestCase(MyTests.TestSOAPFunctions) for test in suite: test.setUp(options.soap_uri) However, I am using xmlrunner for this and its run method takes a TestSuite as an argument. I assume it will run the setUp method itself, so I would need the parameters available within the XMLTestRunner. I hope this makes sense.

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  • String formatting in cocoa

    - by lostInTransit
    Hi I am trying to send some text in an email from my cocoa app (by using Mail.app). Initially I tried using HTML to send properly formatted text. But the mailto: URL does not support html tags (even after setting headers) So I decided to use formatted string (left-aligning of string) This is what I have in my mailto: link's body argument NSMutableString *emailBody = [NSMutableString stringWithFormat:@"|%-35s", [@"Name" UTF8String]]; [emailBody appendFormat:@"|%-18s", [@"Number" UTF8String]]; [emailBody appendString:@"|Notes\n"]; [emailBody appendString:@"----------------------------------------------------------------------------------------------------"]; for(int i = 0; i < [items count]; i++){ NSDictionary *props = [items objectAtIndex:i]; NSMutableString *emailData = [NSMutableString stringWithFormat:@"|%-35s", [[props valueForKey:@"name"] UTF8String]]; [emailData appendFormat:@"|$ %-16s", [[props valueForKey:@"number"] UTF8String]]; [emailData appendString:[props valueForKey:@"notes"]]; [emailBody appendString:@"\n"]; [emailBody appendString:emailData]; } This does give me padded text but they all don't necessarily take up the same space (for instance if there is an O in the text, it takes up more space than others and ruins the formatting) Is there another sure-shot way to format text using just NSString? Thanks

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  • Example of [NSDictionary getObjects:andKeys:]

    - by eagle
    I couldn't find a working example of the method [NSDictionary getObjects:andKeys:] on Google. The only link I could find, doesn't compile. I provided the errors/warnings here in case someone is searching for them. The documentation says that it returns a C array. NSDictionary *myDictionary = ...; id objects[]; // Error: Array size missing in 'objects' id keys[]; // Error: Array size missing in 'keys' [myDictionary getObjects:&objects andKeys:&keys]; for (int i = 0; i < count; i++) { id obj = objects[i]; id key = keys[i]; } . NSDictionary *myDictionary = ...; NSInteger count = [myDictionary count]; id objects[count]; id keys[count]; [myDictionary getObjects:&objects andKeys:&keys]; // Warning: Passing argument 1 of 'getObjects:andKeys:' from incompatible pointer type. for (int i = 0; i < count; i++) { id obj = objects[i]; id key = keys[i]; } Please provide a working example of this method.

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  • "Trying to get property of non-object" error with SimpleXML and PHP

    - by SooDesuNe
    I'm using a PHP script with SimpleXML to parse an XML feed. I have no control over the content of the XML. try { $xml = @new SimpleXMLElement($fetchResult); } catch (Exception $e) { errorHandle($e->getMessage());} $userNick = $xml->View->ScrollView->VBoxView->View->MatrixView->VBoxView[0]->HBoxView->TextView->SetFontStyle->b; foreach ($xml->View->ScrollView->VBoxView->View->MatrixView->VBoxView[0]->VBoxView as $pathToSubTree){ foreach ($pathToSubTree->MatrixView->View->VBoxView->VBoxView->HBoxView[0]->VBoxView->MatrixView->VBoxView as $canopy){ //Do some stuff now that we've found the canopy of the tree } $canopy = $pathToSubTree->MatrixView->View->VBoxView->VBoxView->HBoxView[0]->VBoxView->MatrixView->VBoxView; if(is_null($canopy)){ //Do some stuff stuff is the canopy was not traceable } } $pathToSubTree = $xml->View->ScrollView->VBoxView->View->MatrixView->VBoxView[0]->VBoxView; if(is_null($pathToSubTree)){ //Do some stuff stuff is the subTree path was not traceable } unset($xml); I'm getting lots of two errors, which I'm sure are related to the same cause: PHP Notice: Trying to get property of non-object in myScript.php on line 45 PHP Warning: Invalid argument supplied for foreach() in myScript.php on line 45 PHP Notice: Trying to get property of non-object in myScript.php on line 76 Line 45 is (from above): foreach ($pathToSubTree->MatrixView->View->VBoxView->VBoxView->HBoxView[0]->VBoxView->MatrixView->VBoxView as $canopy){ Line 76 is (from above): $canopy = $pathToSubTree->MatrixView->View->VBoxView->VBoxView->HBoxView[0]->VBoxView->MatrixView->VBoxView; I'm pretty sure this error is caused by one of the arrays in my path not being an array for the particular XML, but some times it CAN be an array. What's the correct way to deal with these?

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  • Calling Matlab's MLApp.MLAppClass.FEval from F#

    - by Matthew
    Matlab provides a COM interface that supports remote execution of arbitrary functions (and code snippets). In particular, it has an Feval method that calls a given Matlab function. The third parameter to this method, pvarArgOut, has COM type VARIANT*, and appears in the Visual Studio F# editor as an argument of type: pvarArgOut: byref<obj> The following code calls interp1, which in Matlab returns a matrix (i.e. 2D double array) result, as is normal for most Matlab functions. let matlab = new MLApp.MLAppClass() let vector_to_array2d (v : vector) = Array2D.init v.Length 1 (fun i j -> v.[i]) let interp1 (xs : vector) (ys : vector) (xi : vector) = let yi : obj = new obj() matlab.Feval("interp1", 1, ref yi, vector_to_array2d xs, vector_to_array2d ys, vector_to_array2d xi) yi :?> float[,] This code compiles fine, but when calling interp1, I get a COM exception: A first chance exception of type 'System.Reflection.TargetInvocationException' occurred in mscorlib.dll A first chance exception of type 'System.Runtime.InteropServices.COMException' occurred in mscorlib.dll An unhandled exception of type 'System.Runtime.InteropServices.COMException' occurred in mscorlib.dll Additional information: Invalid callee. (Exception from HRESULT: 0x80020010 (DISP_E_BADCALLEE)) I get the same error whether initialize yi with a new obj, a new Array2D, or null. How does F# translate VARIANT output arguments?

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  • Writing a Javascript library that is code-completion and code-inspection friendly

    - by Vivin Paliath
    I recently made my own Javascript library and I initially used the following pattern: var myLibrary = (function () { var someProp = "..."; function someFunc() { ... } function someFunc2() { ... } return { func: someFunc, fun2: someFunc2, prop: someProp; } }()); The problem with this is that I can't really use code completion because the IDE doesn't know about the properties that the function literal is returning (I'm using IntelliJ IDEA 9 by the way). I've looked at jQuery code and tried to do this: (function(window, undefined) { var myLibrary = (function () { var someProp = "..."; function someFunc() { ... } function someFunc2() { ... } return { func: someFunc, fun2: someFunc2, prop: someProp; } }()); window.myLibrary = myLibrary; }(window)); I tried this, but now I have a different problem. The IDE doesn't really pick up on myLibrary either. The way I'm solving the problem now is this way: var myLibrary = { func: function() { }, func2: function() { }, prop: "" }; myLibrary = (function () { var someProp = "..."; function someFunc() { ... } function someFunc2() { ... } return { func: someFunc, fun2: someFunc2, prop: someProp; } }()); But that seems kinda clunky, and I can't exactly figure out how jQuery is doing it. Another question I have is how to handle functions with arbitrary numbers of parameters. For example, jQuery.bind can take 2 or 3 parameters, and the IDE doesn't seem to complain. I tried to do the same thing with my library, where a function could take 0 arguments or 1 argument. However, the IDE complains and warns that the correct number of parameters aren't being sent in. How do I handle this?

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  • C# update GUI continuously from backgroundworker.

    - by Qrew
    I have created a GUI (winforms) and added a backgroundworker to run in a separate thread. The backgroundworker needs to update 2 labels continuously. The backgroundworker thread should start with button1 click and run forever. class EcuData { public int RPM { get; set; } public int MAP { get; set; } } private void button1_Click(object sender, EventArgs e) { EcuData data = new EcuData { RPM = 0, MAP = 0 }; BackWorker1.RunWorkerAsync(data); } private void BackWorker1_DoWork(object sender, DoWorkEventArgs e) { EcuData argumentData = e.Argument as EcuData; int x = 0; while (x<=10) { // // Code for reading in data from hardware. // argumentData.RPM = x; //x is for testing only! argumentData.MAP = x * 2; //x is for testing only! e.Result = argumentData; Thread.Sleep(100); x++; } private void BackWorker1_RunWorkerCompleted_1(object sender, RunWorkerCompletedEventArgs e) { EcuData data = e.Result as EcuData; label1.Text = data.RPM.ToString(); label2.Text = data.MAP.ToString(); } } The above code just updated the GUI when backgroundworker is done with his job, and that's not what I'm looking for.

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