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  • "SELECT DISTINCT" ignores different cases

    - by powerbar
    Hello, I have the problem, that MSSQL Server 2000 should select some distinct values from a table (the specific column is of the nvarchar type). There are the sometimes the same values, but with different cases, for example (pseudocode): SELECT DISTINCT * FROM ("A", "a", "b", "B") would return A,b But I do want (and do expect) A,a,b,B because they actually are different values. Any idea how to solve this problem? Thanks a lot in advance!

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  • C++: Return NULL instead of struct

    - by Rosarch
    I have a struct Foo. In pseudocode: def FindFoo: foo = results of search foundFoo = true if a valid foo has been found return foo if foundFoo else someErrorCode How can I accomplish this in C++? Edited to remove numerous inaccuracies.

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  • Command-line video editing in Linux (cut, join and preview)

    - by sdaau
    I have rather simple editing needs - I need to cut up some videos, maybe insert some PNGs in between them, and join these videos (don't need transitions, effects, etc.). Basically, pitivi would do what I want - except, I use 640x480 30 fps AVI's from a camera, and as soon as I put in over a couple of minutes of that kind of material, pitivi starts freezing on preview, and thus becomes unusable. So, I started looking for a command line tool for Linux; I guess only ffmpeg (command line - Using ffmpeg to cut up video - Super User) and mplayer (Sam - Edit video file with mencoder under linux) are so far candidates, but I cannot find examples of the use I have in mind.   Basically, I'd imagine there's an encoder and player tools (like ffmpeg vs ffplay; or mencoder vs mplayer) - such that, to begin with, the edit sequence could be specified directly on the command line, preferably with frame resolution - a pseudocode would look like: videnctool -compose --file=vid1.avi --start=00:00:30:12 --end=00:01:45:00 --file=vid2.avi --start=00:05:00:00 --end=00:07:12:25 --file=mypicture.png --duration=00:00:02:00 --file=vid3.avi --start=00:02:00:00 --end=00:02:45:10 --output=editedvid.avi ... or, it could have a "playlist" text file, like: vid1.avi 00:00:30:12 00:01:45:00 vid2.avi 00:05:00:00 00:07:12:25 mypicture.png - 00:00:02:00 vid3.avi 00:02:00:00 00:02:45:10 ... so it could be called with videnctool -compose --playlist=playlist.txt --output=editedvid.avi The idea here would be that all of the videos are in the same format - allowing the tool to avoid transcoding, and just do a "raw copy" instead (as in mencoder's copy codec: "-oac copy -ovc copy") - or in lack of that, uncompressed audio/video would be OK (although it would eat a bit of space). In the case of the still image, the tool would use the encoding set by the video files.   The thing is, I can so far see that mencoder and ffmpeg can operate on individual files; e.g. cut a single section from a single file, or join files (mencoder also has Edit Decision Lists (EDL), which can be used to do frame-exact cutting - so you can define multiple cut regions, but it's again attributed to a single file). Which implies I have to work on cutting pieces first from individual files first (each of which would demand own temporary file on disk), and then joining them in a final video file. I would then imagine, that there is a corresponding player tool, which can read the same command line option format / playlist file as the encoding tool - except it will not generate an output file, but instead play the video; e.g. in pseudocode: vidplaytool --playlist=playlist.txt --start=00:01:14 --end=00:03:13 ... and, given there's enough memory, it would generate a low-res video preview in RAM, and play it back in a window, while offering some limited interaction ( like mplayer's keyboard shortcuts for play, pause, rewind, step frame). Of course, I'd imagine the start and end times to refer to the entire playlist, and include any file that may end up in that region in the playlist. Thus, the end result of all this would be: command line operation; no temporary files while doing the editing - and also no temporary files (nor transcoding) when rendering final output... which I myself think would be nice. So, while I think that all of the above may be a bit of a stretch - does there exist anything that would approximate the workflow described above?

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  • Failover tmpfs mirroring. Am I doing it right?

    - by user45286
    My goal is to have a certain directory to be available as tmpfs. There will be some modifications during server uptime in this dir and those modifications must be synced to non-tmpfs persistent dir on HDD over rsync. After server boot the latest version from non-tmpfs persistent dir must be moved to tmpfs and rsync syncing to be started. I'm afraid that rsync will erase non-tmpfs backup if tmpfs dir will be empty.. I'm doing it in this way right now: create tmpfs partition in /etc/fstab cat /etc/rc.local (pseudocode) delete "tmpfs rsync" cronjob from /var/spool/cron/crontabs if there is any cp -r /path/to/non-tmpfs-backup /path/to/tmpfs/dir append /var/spool/cron/crontabs with "tmpfs rsync" cronjob What do you think?

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  • How to automate changing my ip?

    - by callisto
    I am very new to OSX. I will use my MBP at work and home. I would like to be easily able to switch my ip when changing location. Thus far I have dabbled with the automator, hoping to do something like this: [pseudocode] If IP = 192.168.0.10 root# changeip 192.168.0.10 10.0.0.15 else root# changeip 10.0.0.15 192.168.0.10 The reason for this is that my IP from home will not allow me access at work and vice versa. I have friends and family who drop in now and then, multiple wireless devices set up for the home IP range. Changing all of that to accommodate one new device (the Macbook) would make me reconsider my foray into OSX. I'd rather have the MBP adapt to me than I to it.

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  • Recursive Batch File

    - by MCZ
    I have a file that looks this: head1,head2,head3,head4,head5,head6 a11,a12,keyA,a14,a15,a16 a21,a22,keyB,a24,a25 a31,a32,keyC,a34 a41,a42,keyB,a44,a44 a51,a52,keyA,a54,a55,a56 a61,a62,keyA,a64,a65,a66 a71,a72,keyC,a74 some message Objective: Write list of unique keys to a text file. For example, the result for the file described above should be: keyA, keyB, keyC Here's the pseudocode I would like to implement in batch file recur.bat Read second line of inputfile If no key exist on second line, return else continue Append keyX to list FINDSTR /v keyX inputfile Pipe results to recur.bat I don't know if this is the most efficient way to do this without using actual programming language. Any suggestions for actual batch file code?

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  • load syntax as per file prefix

    - by Richo
    Firstly, I hope that this is the right place. I couldn't decide between here and superuser. My home directory lives in an svn repo. all my dotfiles are in version control so that I can track them across multiple machines, and they all source an unversioned .local (ie, .screenrc.local, .vimrc.local etc) which can override/make local changes to the environment in a machine specific way. The problem is that vim understands how I want to edit many of these config files, but loses it's mind when I open a .local, and honestly, I'm not really sure what it does to work out how to syntax highlight etc a file like .screenrc the pseudocode for what I'm after is: if OpenedFile.ends_with(".local") behave_as_per OpenedFile[0:-6] endif I hope this makes sense and hopefully someone can shed light on whether or not this is possible.

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  • Collision Detection Code Structure with Sloped Tiles

    - by ProgrammerGuy123
    Im making a 2D tile based game with slopes, and I need help on the collision detection. This question is not about determining the vertical position of the player given the horizontal position when on a slope, but rather the structure of the code. Here is my pseudocode for the collision detection: void Player::handleTileCollisions() { int left = //find tile that's left of player int right = //find tile that's right of player int top = //find tile that's above player int bottom = //find tile that's below player for(int x = left; x <= right; x++) { for(int y = top; y <= bottom; y++) { switch(getTileType(x, y)) { case 1: //solid tile { //resolve collisions break; } case 2: //sloped tile { //resolve collisions break; } default: //air tile or whatever else break; } } } } When the player is on a sloped tile, he is actually inside the tile itself horizontally, that way the player doesn't look like he is floating. This creates a problem because when there is a sloped tile next to a solid square tile, the player can't move passed it because this algorithm resolves any collisions with the solid tile. Here is a gif showing this problem: So what is a good way to structure my code so that when the player is inside a sloped tile, solid tiles get ignored?

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  • Triangulation A* (TA*) pathfinding algorithm

    - by hyn
    I need help understanding the Triangle A* (TA*) algorithm that is described by Demyen in his paper Efficient Triangulation-Based Pathfinding, on pages 76-81. He describes how to adapt the regular A* algorithm for triangulation, to search for other possibly more optimal paths, even after the final node is reached/expanded. Regular A* stops when the final node is expanded, but this is not always the best path when used in a triangulated graph. This is exactly the problem I'm having. The problem is illustrated on page 78, Figure 5.4: I understand how to calculate the g and h values presented in the paper (page 80). And I think the search stop condition is: if (currentNode.fCost > shortestDistanceFound) { // stop break; } where currentNode is the search node popped from the open list (priority queue), which has the lowest f-score. shortestDistanceFound is the actual distance of the shortest path found so far. But how do I exclude the previously found paths from future searches? Because if I do the search again, it will obviously find the same path. Do I reset the closed list? I need to modify something, but I don't know what it is I need to change. The paper lacks pseudocode, so that would be helpful.

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  • Allocation algorithm help, using Python.

    - by Az
    Hi there, I've been working on this general allocation algorithm for students. The pseudocode for it (a Python implementation) is: for a student in a dictionary of students: for student's preference in a set of preferences (ordered from 1 to 10): let temp_project be the first preferred project check if temp_project is available if so, allocate it to them and make the project UNavailable to others Quite simply this will try to allocate projects by starting from their most preferred. The way it works, out of a set of say 100 projects, you list 10 you would want to do. So the 10th project wouldn't be the "least preferred overall" but rather the least preferred in their chosen set, which isn't so bad. Obviously if it can't allocate a project, a student just reverts to the base case which is an allocation of None, with a rank of 11. What I'm doing is calculating the allocation "quality" based on a weighted sum of the ranks. So the lower the numbers (i.e. more highly preferred projects), the better the allocation quality (i.e. more students have highly preferred projects). That's basically what I've currently got. Simple and it works. Now I'm working on this algorithm that tries to minimise the allocation weight locally (this pseudocode is a bit messy, sorry). The only reason this will probably work is because my "search space" as it is, isn't particularly large (just a very general, anecdotal observation, mind you). Since the project is only specific to my Department, we have their own limits imposed. So the number of students can't exceed 100 and the number of preferences won't exceed 10. for student in a dictionary/list/whatever of students: where i = 0 take the (i)st student, (i+1)nd student for their ranks: allocate the projects and set local_weighting to be sum(student_i.alloc_proj_rank, student_i+1.alloc_proj_rank) these are the cases: if local_weighting is 2 (i.e. both ranks are 1): then i += 1 and and continue above if local weighting is = N>2 (i.e. one or more ranks are greater than 1): let temp_local_weighting be N: pick student with lowest rank and then move him to his next rank and pick the other student and reallocate his project after this if temp_local_weighting is < N: then allocate those projects to the students move student with lowest rank to the next rank and reallocate other if temp_local_weighting < previous_temp_allocation: let these be the new allocated projects try moving for the lowest rank and reallocate other else: if this weighting => previous_weighting let these be the allocated projects i += 1 and move on for the rest of the students So, questions: This is sort of a modification of simulated annealing, but any sort of comments on this would be appreciated. How would I keep track of which student is (i) and which student is (i+1) If my overall list of students is 100, then the thing would mess up on (i+1) = 101 since there is none. How can I circumvent that? Any immediate flaws that can be spotted? Extra info: My students dictionary is designed as such: students[student_id] = Student(student_id, student_name, alloc_proj, alloc_proj_rank, preferences) where preferences is in the form of a dictionary such that preferences[rank] = {project_id}

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  • Writing more efficient xquery code (avoiding redundant iteration)

    - by Coquelicot
    Here's a simplified version of a problem I'm working on: I have a bunch of xml data that encodes information about people. Each person is uniquely identified by an 'id' attribute, but they may go by many names. For example, in one document, I might find <person id=1>Paul Mcartney</person> <person id=2>Ringo Starr</person> And in another I might find: <person id=1>Sir Paul McCartney</person> <person id=2>Richard Starkey</person> I want to use xquery to produce a new document that lists every name associated with a given id. i.e.: <person id=1> <name>Paul McCartney</name> <name>Sir Paul McCartney</name> <name>James Paul McCartney</name> </person> <person id=2> ... </person> The way I'm doing this now in xquery is something like this (pseudocode-esque): let $ids := distinct-terms( [all the id attributes on people] ) for $id in $ids return <person id={$id}> { for $unique-name in distinct-values ( for $name in ( [all names] ) where $name/@id=$id return $name ) return <name>{$unique-name}</name> } </person> The problem is that this is really slow. I imagine the bottleneck is the innermost loop, which executes once for every id (of which there are about 1200). I'm dealing with a fair bit of data (300 MB, spread over about 800 xml files), so even a single execution of the query in the inner loop takes about 12 seconds, which means that repeating it 1200 times will take about 4 hours (which might be optimistic - the process has been running for 3 hours so far). Not only is it slow, it's using a whole lot of virtual memory. I'm using Saxon, and I had to set java's maximum heap size to 10 GB (!) to avoid getting out of memory errors, and it's currently using 6 GB of physical memory. So here's how I'd really like to do this (in Pythonic pseudocode): persons = {} for id in ids: person[id] = set() for person in all_the_people_in_my_xml_document: persons[person.id].add(person.name) There, I just did it in linear time, with only one sweep of the xml document. Now, is there some way to do something similar in xquery? Surely if I can imagine it, a reasonable programming language should be able to do it (he said quixotically). The problem, I suppose, is that unlike Python, xquery doesn't (as far as I know) have anything like an associative array. Is there some clever way around this? Failing that, is there something better than xquery that I might use to accomplish my goal? Because really, the computational resources I'm throwing at this relatively simple problem are kind of ridiculous.

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  • How does flocking algorithm work?

    - by Chan
    I read and understand the basic of flocking algorithm. Basically, we need to have 3 behaviors: 1. Cohesion 2. Separation 3. Alignment From my understanding, it's like a state machine. Every time we do an update (then draw), we check all the constraints on both three behaviors. And each behavior returns a Vector3 which is the "correct" orientation that an object should transform to. So my initial idea was /// <summary> /// Objects stick together /// </summary> /// <returns></returns> private Vector3 Cohesion() { Vector3 result = new Vector3(0.0f, 0.0f, 0.0f); return result; } /// <summary> /// Object align /// </summary> /// <returns></returns> private Vector3 Align() { Vector3 result = new Vector3(0.0f, 0.0f, 0.0f); return result; } /// <summary> /// Object separates from each others /// </summary> /// <returns></returns> private Vector3 Separate() { Vector3 result = new Vector3(0.0f, 0.0f, 0.0f); return result; } Then I search online for pseudocode but many of them involve velocity and acceleration plus other stuffs. This part confused me. In my game, all objects move at constant speed, and they have one leader. So can anyone share me an idea how to start on implement this flocking algorithm? Also, did I understand it correctly? (I'm using XNA 4.0)

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  • Multiple audio sources on a single gameObject in unity

    - by angryInsomniac
    So, I have an audio system set up wherein I have loaded all my audio clips centrally and play them on demand by passing the requesting audioSource into the sound manager. However, there is a complication wherein if I want to overlay multiple looping sounds, I need to have multiple audio sources on an object, which is fine , so I created two in my script instantiated them and played my clips on them and then the world went crazy. For some reason, when I create two audio Sources in an object only the latest one is ever used, even if I explicitly keep objects separated, playing a clip on one or the other plays the clip on the last one that was created, furthermore, either this last one is not created in the right place or somehow messes with the rolloff rules because I can hear it all across my level, havign just one source works fine, but putting a second one on it causes shit to go batshit insane. Does anyone know the reason / solution for this ? Some pseudocode : guardSoundsSource = (AudioSource)gameObject.AddComponent("AudioSource"); guardSoundsSource.name = "Guard_Sounds_source"; // Setup this source guardThrusterSource = (AudioSource)gameObject.AddComponent("AudioSource"); guardThrusterSource.name = "Guard_Thruster_Source"; // setup this source // play using custom Sound manager soundMan.soundMgr.playOnSource(guardSoundsSource,"Guard_Idle_loop" ,true,GameManager.Manager.PlayerType); // this method prints out the name of the source the sound was to be played on and it always shows "Guard_Thruster_Source" even on the "Guard_Idle_loop" even though I clearly told it to use "Guard_Sounds_source"

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  • Making a 2D game with responsive resolution

    - by alexandervrs
    I am making a 2D game, however I wish for it to be resolution agnostic. My target resolution i.e. where things look as intended is 1600 x 900. My ideas are: Make the HUD stay fixed to the sides no matter what resolution, use different size for HUD graphics under a certain resolution and another under a certain large one. Use large HD PNG sprites/backgrounds which are a power of 2, so they scale nicely. No vectors. Use the player's native resolution. Scale the game area (not the HUD) to fit (resulting zooming in some and cropping the game area sides if necessary for widescreen, no stretch), but always fill the screen. Have a min and max resolution limit for small and very large displays where you will just change the resolution(?) or scale up/down to fit. What I am a bit confused though is what math formula I would use to scale the game area correctly based on the resolution no matter the aspect ratio, fully fit in a square screen and with some clip to the sides for widescreen. Pseudocode would help as well. :)

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  • How should I architect my Model and Data Access layer objects in my website?

    - by Robin Winslow
    I've been tasked with designing Data layer for a website at work, and I am very interested in architecture of code for the best flexibility, maintainability and readability. I am generally acutely aware of the value in completely separating out my actual Models from the Data Access layer, so that the Models are completely naive when it comes to Data Access. And in this case it's particularly useful to do this as the Models may be built from the Database or may be built from a Soap web service. So it seems to me to make sense to have Factories in my data access layer which create Model objects. So here's what I have so far (in my made-up pseudocode): class DataAccess.ProductsFromXml extends DataAccess.ProductFactory {} class DataAccess.ProductsFromDatabase extends DataAccess.ProductFactory {} These then get used in the controller in a fashion similar to the following: var xmlProductCreator = DataAccess.ProductsFromXml(xmlDataProvider); var databaseProductCreator = DataAccess.ProductsFromXml(xmlDataProvider); // Returns array of Product model objects var XmlProducts = databaseProductCreator.Products(); // Returns array of Product model objects var DbProducts = xmlProductCreator.Products(); So my question is, is this a good structure for my Data Access layer? Is it a good idea to use a Factory for building my Model objects from the data? Do you think I've misunderstood something? And are there any general patterns I should read up on for how to write my data access objects to create my Model objects?

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  • In an Entity/Component system, can component data be implemented as a simple array of key-value pairs? [on hold]

    - by 010110110101
    I'm trying to wrap my head around how to organize components in an Entity Component Systems once everything in the current scene/level is loaded in memory. (I'm a hobbyist BTW) Some people seem to implement the Entity as an object that contains a list of of "Component" objects. Components contain data organized as an array of key-value pairs. Where the value is serialized "somehow". (pseudocode is loosely in C# for brevity) class Entity { Guid _id; List<Component> _components; } class Component { List<ComponentAttributeValue> _attributes; } class ComponentAttributeValue { string AttributeName; object AttributeValue; } Others describe Components as an in-memory "table". An entity acquires the component by having its key placed in a table. The attributes of the component-entity instance are like the columns in a table class Renderable_Component { List<RenderableComponentAttributeValue> _entities; } class RenderableComponentAttributeValue { Guid entityId; matrix4 transformation; // other stuff for rendering // everything is strongly typed } Others describe this actually as a table. (and such tables sound like an EAV database schema BTW) (and the value is serialized "somehow") Render_Component_Table ---------------- Entity Id Attribute Name Attribute Value and when brought into running code: class Entity { Guid _id; Dictionary<string, object> _attributes; } My specific question is: Given various components, (Renderable, Positionable, Explodeable, Hideable, etc) and given that each component has an attribute with a particular name, (TRANSLATION_MATRIX, PARTICLE_EMISSION_VELOCITY, CAN_HIDE, FAVORITE_COLOR, etc) should: an entity contain a list of components where each component, in turn, has their own array of named attributes with values serialized somehow or should components exist as in-memory tables of entity references and associated with each "row" there are "columns" representing the attribute with values that are specific to each entity instance and are strongly typed or all attributes be stored in an entity as a singular array of named attributes with values serialized somehow (could have name collisions) or something else???

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  • Making a game with responsive resolution

    - by alexandervrs
    I am making a game, however I wish for it to be resolution agnostic. My target resolution i.e. where things look as intended is 1600 x 900. My ideas are: Make the HUD stay fixed to the sides no matter what resolution, use different size for HUD graphics under a certain resolution and another under a certain large one. Use large HD sprites/backgrounds which are a power of 2, so they scale nicely. Use the player's native resolution. Scale the game area (not the HUD) to fit (resulting zooming in some and cropping the game area sides if necessary for widescreen, no stretch), but always fill the screen. Have a min and max resolution limit for small and very large displays where you will just change the resolution(?) or scale up/down to fit. What I am a bit confused though is what math formula I would use to scale the game area correctly based on the resolution no matter the aspect ratio, fully fit in a square screen and with some clip to the sides for widescreen. Pseudocode would help as well. :)

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  • [C#][XNA] Draw() 20,000 32 by 32 Textures or 1 Large Texture 20,000 Times

    - by Rudi
    The title may be confusing - sorry about that, it's a poor summary. Here's my dilemma. I'm programming in C# using the .NET Framework 4, and aiming to make a tile-based game with XNA. I have one large texture (256 pixels by 4096 pixels). Remember this is a tile-based game, so this texture is so massive only because it contains many tiles, which are each 32 pixels by 32 pixels. I think the experts will definitely know what a tile-based game is like. The orientation is orthogonal (like a chess board), not isometric. In the Game.Draw() method, I have two choices, one of which will be incredibly more efficient than the other. Choice/Method #1: Semi-Pseudocode: public void Draw() { // map tiles are drawn left-to-right, top-to-bottom for (int x = 0; x < mapWidth; x++) { for (int y = 0; y < mapHeight; y++) { SpriteBatch.Draw( MyLargeTexture, // One large 256 x 4096 texture new Rectangle(x, y, 32, 32), // Destination rectangle - ignore this, its ok new Rectangle(x, y, 32, 32), // Notice the source rectangle 'cuts out' 32 by 32 squares from the texture corresponding to the loop Color.White); // No tint - ignore this, its ok } } } Caption: So, effectively, the first method is referencing one large texture many many times, each time using a small rectangle of this large texture to draw the appropriate tile image. Choice/Method #2: Semi-Pseudocode: public void Draw() { // map tiles are drawn left-to-right, top-to-bottom for (int x = 0; x < mapWidth; x++) { for (int y = 0; y < mapHeight; y++) { Texture2D tileTexture = map.GetTileTexture(x, y); // Getting a small 32 by 32 texture (different each iteration of the loop) SpriteBatch.Draw( tileTexture, new Rectangle(x, y, 32, 32), // Destination rectangle - ignore this, its ok new Rectangle(0, 0, tileTexture.Width, tileTexture.Height), // Notice the source rectangle uses the entire texture, because the entire texture IS 32 by 32 Color.White); // No tint - ignore this, its ok } } } Caption: So, effectively, the second method is drawing many small textures many times. The Question: Which method and why? Personally, I would think it would be incredibly more efficient to use the first method. If you think about what that means for the tile array in a map (think of a large map with 2000 by 2000 tiles, let's say), each Tile object would only have to contain 2 integers, for the X and Y positions of the source rectangle in the one large texture - 8 bytes. If you use method #2, however, each Tile object in the tile array of the map would have to store a 32by32 Texture - an image - which has to allocate memory for the R G B A pixels 32 by 32 times - is that 4096 bytes per tile then? So, which method and why? First priority is speed, then memory-load, then efficiency or whatever you experts believe.

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  • Prefer class members or passing arguments between internal methods?

    - by geoffjentry
    Suppose within the private portion of a class there is a value which is utilized by multiple private methods. Do people prefer having this defined as a member variable for the class or passing it as an argument to each of the methods - and why? On one hand I could see an argument to be made that reducing state (ie member variables) in a class is generally a good thing, although if the same value is being repeatedly used throughout a class' methods it seems like that would be an ideal candidate for representation as state for the class to make the code visibly cleaner if nothing else. Edit: To clarify some of the comments/questions that were raised, I'm not talking about constants and this isn't relating to any particular case rather just a hypothetical that I was talking to some other people about. Ignoring the OOP angle for a moment, the particular use case that I had in mind was the following (assume pass by reference just to make the pseudocode cleaner) int x doSomething(x) doAnotherThing(x) doYetAnotherThing(x) doSomethingElse(x) So what I mean is that there's some variable that is common between multiple functions - in the case I had in mind it was due to chaining of smaller functions. In an OOP system, if these were all methods of a class (say due to refactoring via extracting methods from a large method), that variable could be passed around them all or it could be a class member.

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  • Problem with a* implementation in pygame

    - by piyush3dxyz
    Yesterday i decide to make RTS game in pygame(pygame is best).I figured out many components of RTS game like unit selecting,health,resources but only 1 thing i still not understand.. which is a* pathfinding in pygame... I also done little bit of research on wiki,articles and papers...but still cant figure out problem.... function A*(start,goal) closedset := the empty set // The set of nodes already evaluated. openset := {start} // The set of tentative nodes to be evaluated, initially containing the start node came_from := the empty map // The map of navigated nodes. g_score[start] := 0 // Cost from start along best known path. // Estimated total cost from start to goal through y. f_score[start] := g_score[start] + heuristic_cost_estimate(start, goal) while openset is not empty current := the node in openset having the lowest f_score[] value if current = goal return reconstruct_path(came_from, goal) remove current from openset add current to closedset for each neighbor in neighbor_nodes(current) if neighbor in closedset continue tentative_g_score := g_score[current] + dist_between(current,neighbor) if neighbor not in openset or tentative_g_score <= g_score[neighbor] came_from[neighbor] := current g_score[neighbor] := tentative_g_score f_score[neighbor] := g_score[neighbor] + heuristic_cost_estimate(neighbor, goal) if neighbor not in openset add neighbor to openset return failure here is the pseudocode for wiki a* implementation......

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  • Where ORMs blur the lines between code and data, how do you decide what logic should be a stored procedure, and what should be coded?

    - by PhonicUK
    Take the following pseudocode: CreateInvoiceAndCalculate(ItemsAndQuantities, DispatchAddress, User); And say CreateInvoice does the following: Create a new entry in an Invoices table belonging to the specified User to be sent to the given DispatchAddress. Create a new entry in an InvoiceItems table for each of the items in ItemsAndQuantities, storing the Item, the Quantity, and the cost of the item as of now (by looking it up from an Items table) Calculate the total amount of the invoice (ex shipping and taxes) and store it in the new Invoice row. At a glace you wouldn't be able to tell if this was a method in my applications code, or a stored procedure in the database that is being exposed as a function by the ORM. And to some extent it doesn't really matter. Now technically none of this is business logic. You're not making any decisions - just performing a calculation and creating records. However some may argue that because you are performing a calculation that affects the business (the total amount to be invoiced) that this isn't something that should be done in a stored procedure and instead should be in code. So for this specific example - why would it be more appropriate to do one or the other? And where do you draw the line? Or does it even particular matter as long as it's sufficiently well documented?

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  • hosting simple python scripts in a container to handle concurrency, configuration, caching, etc.

    - by Justin Grant
    My first real-world Python project is to write a simple framework (or re-use/adapt an existing one) which can wrap small python scripts (which are used to gather custom data for a monitoring tool) with a "container" to handle boilerplate tasks like: fetching a script's configuration from a file (and keeping that info up to date if the file changes and handle decryption of sensitive config data) running multiple instances of the same script in different threads instead of spinning up a new process for each one expose an API for caching expensive data and storing persistent state from one script invocation to the next Today, script authors must handle the issues above, which usually means that most script authors don't handle them correctly, causing bugs and performance problems. In addition to avoiding bugs, we want a solution which lowers the bar to create and maintain scripts, especially given that many script authors may not be trained programmers. Below are examples of the API I've been thinking of, and which I'm looking to get your feedback about. A scripter would need to build a single method which takes (as input) the configuration that the script needs to do its job, and either returns a python object or calls a method to stream back data in chunks. Optionally, a scripter could supply methods to handle startup and/or shutdown tasks. HTTP-fetching script example (in pseudocode, omitting the actual data-fetching details to focus on the container's API): def run (config, context, cache) : results = http_library_call (config.url, config.http_method, config.username, config.password, ...) return { html : results.html, status_code : results.status, headers : results.response_headers } def init(config, context, cache) : config.max_threads = 20 # up to 20 URLs at one time (per process) config.max_processes = 3 # launch up to 3 concurrent processes config.keepalive = 1200 # keep process alive for 10 mins without another call config.process_recycle.requests = 1000 # restart the process every 1000 requests (to avoid leaks) config.kill_timeout = 600 # kill the process if any call lasts longer than 10 minutes Database-data fetching script example might look like this (in pseudocode): def run (config, context, cache) : expensive = context.cache["something_expensive"] for record in db_library_call (expensive, context.checkpoint, config.connection_string) : context.log (record, "logDate") # log all properties, optionally specify name of timestamp property last_date = record["logDate"] context.checkpoint = last_date # persistent checkpoint, used next time through def init(config, context, cache) : cache["something_expensive"] = get_expensive_thing() def shutdown(config, context, cache) : expensive = cache["something_expensive"] expensive.release_me() Is this API appropriately "pythonic", or are there things I should do to make this more natural to the Python scripter? (I'm more familiar with building C++/C#/Java APIs so I suspect I'm missing useful Python idioms.) Specific questions: is it natural to pass a "config" object into a method and ask the callee to set various configuration options? Or is there another preferred way to do this? when a callee needs to stream data back to its caller, is a method like context.log() (see above) appropriate, or should I be using yield instead? (yeild seems natural, but I worry it'd be over the head of most scripters) My approach requires scripts to define functions with predefined names (e.g. "run", "init", "shutdown"). Is this a good way to do it? If not, what other mechanism would be more natural? I'm passing the same config, context, cache parameters into every method. Would it be better to use a single "context" parameter instead? Would it be better to use global variables instead? Finally, are there existing libraries you'd recommend to make this kind of simple "script-running container" easier to write?

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  • Is there an alternative to the term "calling object"?

    - by ybakos
    Let's suppose you've got a class defined (in pseudocode): class Puppy { // ... string sound = "Rawr!"; void bark() { print(sound); } } And say, given a Puppy instance, you call it's bark() method: Puppy p; p.bark(); Notice how bark() uses the member variable sound. In many contexts, I've seen folks describe sound as the member variable of the "calling object." My question is, what's a better term to use than "calling object?" To me, the object is not doing any calling. We know that member functions are in a way just functions with an implicit this or self parameter. I've come up with "receiving object," or "message recipient," which makes sense if you're down with the "messaging" paradigm. Do any of you happy hackers have a term that you like to use? I feel it should mean "the object upon which a method is called" and TOUWAMIC just doesn't cut it.

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  • Handling commands or events that wait for an action to be completed afterwards

    - by virulent
    Say you have two events: Action1 and Action2. When you receive Action1, you want to store some arbitrary data to be used the next time Action2 rolls around. Optimally, Action1 is normally a command however it can also be other events. The idea is still the same. The current way I am implementing this is by storing state and then simply checking when Action2 is called if that specific state is there. This is obviously a bit messy and leads to a lot of redundant code. Here is an example of how I am doing that, in pseudocode form (and broken down quite a bit, obviously): void onAction1(event) { Player = event.getPlayer() Player.addState("my_action_to_do") } void onAction2(event) { Player = event.getPlayer() if not Player.hasState("my_action_to_do") { return } // Do something } When doing this for a lot of other actions it gets somewhat ugly and I wanted to know if there is something I can do to improve upon it. I was thinking of something like this, which wouldn't require passing data around, but is this also not the right direction? void onAction1(event) { Player = event.getPlayer() Player.onAction2(new Runnable() { public void run() { // Do something } }) } If one wanted to take it even further, could you not simply do this? void onPlayerEnter(event) { // When they join the server Player = event.getPlayer() Player.onAction1(new Runnable() { public void run() { // Now wait for action 2 Player.onAction2(new Runnable() { // Do something }) } }, true) // TRUE would be to repeat the event, // not remove it after it is called. } Any input would be wonderful.

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  • How to structure reading of commands given at a(n interactive) CLI prompt?

    - by Anto
    Let's say I have a program called theprogram (the marketing team was on strike when the product was to be named). I start that program by typing, perhaps not surprisingly, the program name as a command into a command prompt. After that, I get into a loop (from the users standpoint, an interactive command-line prompt), where one command will be read from the user, and depending on what command was given, the program will execute some instructions. I have been doing something like the following (in C-like pseudocode): main_loop{ in=read_input(); if(in=="command 1") do_something(); else if(in=="command 2") do_something_else(); ... } (In a real program, I would probably encapsulate more things into different procedures, this is just an example.) This works well for a small amount of commands, but let's say you have 100, 1000 or even 10 000 of them (the manual would be huge!). It is clearly a bad idea to have 10 000 ifs and else ifs after each other, for instance, the program would be hard to read, hard to maintain, contain a lot of boilerplate code... Yeah, you don't want to do that, so what approach would you recommend me to use (I will probably never use 10 000 commands in a program, but the solution should, at least preferably, be able to scale to that kind of massive (?) problems. The solution doesn't have to allow for arguments to the commands)?

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