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• volume group disappeared after xfs_check run

  - by John P

  EDIT** I have a volume group consisting of 5 RAID1 devices grouped together into a lvm and formatted with xfs. The 5th RAID device lost its RAID config (cat /proc/mdstat does not show anything). The two drives are still present (sdj and sdk), but they have no partitions. The LVM appeared to be happily using sdj up until recently. (doing a pvscan showed the first 4 RAID1 devices + /dev/sdj) I removed the LVM from the fstab, rebooted, then ran xfs_check on the LV. It ran for about half an hour, then stopped with an error. I tried rebooting again, and this time when it came up, the logical volume was no longer there. It is now looking for /dev/md5, which is gone (though it had been using /dev/sdj earlier). /dev/sdj was having read errors, but after replacing the SATA cable, those went away, so the drive appears to be fine for now. Can I modify the /etc/lvm/backup/dedvol, change the device to /dev/sdj and do a vgcfgrestore? I could try doing a pvcreate --uuid KZron2-pPTr-ZYeQ-PKXX-4Woq-6aNc-AG4rRJ /dev/sdj to make it recognize it, but I'm afraid that would erase the data on the drive UPDATE: just changing the pv to point to /dev/sdj did not work vgcfgrestore --file /etc/lvm/backup/dedvol dedvol Couldn't find device with uuid 'KZron2-pPTr-ZYeQ-PKXX-4Woq-6aNc-AG4rRJ'. Cannot restore Volume Group dedvol with 1 PVs marked as missing. Restore failed. pvscan /dev/sdj: read failed after 0 of 4096 at 0: Input/output error Couldn't find device with uuid 'KZron2-pPTr-ZYeQ-PKXX-4Woq-6aNc-AG4rRJ'. Couldn't find device with uuid 'KZron2-pPTr-ZYeQ-PKXX-4Woq-6aNc-AG4rRJ'. Couldn't find device with uuid 'KZron2-pPTr-ZYeQ-PKXX-4Woq-6aNc-AG4rRJ'. Couldn't find device with uuid 'KZron2-pPTr-ZYeQ-PKXX-4Woq-6aNc-AG4rRJ'. PV /dev/sdd2 VG VolGroup00 lvm2 [74.41 GB / 0 free] PV /dev/md2 VG dedvol lvm2 [931.51 GB / 0 free] PV /dev/md3 VG dedvol lvm2 [931.51 GB / 0 free] PV /dev/md0 VG dedvol lvm2 [931.51 GB / 0 free] PV /dev/md4 VG dedvol lvm2 [931.51 GB / 0 free] PV unknown device VG dedvol lvm2 [1.82 TB / 63.05 GB free] Total: 6 [5.53 TB] / in use: 6 [5.53 TB] / in no VG: 0 [0 ] vgscan Reading all physical volumes. This may take a while... /dev/sdj: read failed after 0 of 4096 at 0: Input/output error /dev/sdj: read failed after 0 of 4096 at 2000398843904: Input/output error Found volume group "VolGroup00" using metadata type lvm2 Found volume group "dedvol" using metadata type lvm2 vgdisplay dedvol --- Volume group --- VG Name dedvol System ID Format lvm2 Metadata Areas 5 Metadata Sequence No 10 VG Access read/write VG Status resizable MAX LV 0 Cur LV 1 Open LV 0 Max PV 0 Cur PV 5 Act PV 5 VG Size 5.46 TB PE Size 4.00 MB Total PE 1430796 Alloc PE / Size 1414656 / 5.40 TB Free PE / Size 16140 / 63.05 GB VG UUID o1U6Ll-5WH8-Pv7Z-Rtc4-1qYp-oiWA-cPD246 dedvol { id = "o1U6Ll-5WH8-Pv7Z-Rtc4-1qYp-oiWA-cPD246" seqno = 10 status = ["RESIZEABLE", "READ", "WRITE"] flags = [] extent_size = 8192 # 4 Megabytes max_lv = 0 max_pv = 0 physical_volumes { pv0 { id = "Msiee7-Zovu-VSJ3-Y2hR-uBVd-6PaT-Ho9v95" device = "/dev/md2" # Hint only status = ["ALLOCATABLE"] flags = [] dev_size = 1953519872 # 931.511 Gigabytes pe_start = 384 pe_count = 238466 # 931.508 Gigabytes } pv1 { id = "ZittCN-0x6L-cOsW-v1v4-atVN-fEWF-e3lqUe" device = "/dev/md3" # Hint only status = ["ALLOCATABLE"] flags = [] dev_size = 1953519872 # 931.511 Gigabytes pe_start = 384 pe_count = 238466 # 931.508 Gigabytes } pv2 { id = "NRNo0w-kgGr-dUxA-mWnl-bU5v-Wld0-XeKVLD" device = "/dev/md0" # Hint only status = ["ALLOCATABLE"] flags = [] dev_size = 1953519872 # 931.511 Gigabytes pe_start = 384 pe_count = 238466 # 931.508 Gigabytes } pv3 { id = "2EfLFr-JcRe-MusW-mfAs-WCct-u4iV-W0pmG3" device = "/dev/md4" # Hint only status = ["ALLOCATABLE"] flags = [] dev_size = 1953519872 # 931.511 Gigabytes pe_start = 384 pe_count = 238466 # 931.508 Gigabytes } pv4 { id = "KZron2-pPTr-ZYeQ-PKXX-4Woq-6aNc-AG4rRJ" device = "/dev/md5" # Hint only status = ["ALLOCATABLE"] flags = [] dev_size = 3907028992 # 1.81935 Terabytes pe_start = 384 pe_count = 476932 # 1.81935 Terabytes } }

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• Yet another frustum culling question

  - by Christian Frantz

  This one is kinda specific. If I'm to implement frustum culling in my game, that means each one of my cubes would need a bounding sphere. My first question is can I make the sphere so close to the edge of the cube that its still easily clickable for destroying and building? Frustum culling is easily done in XNA as I've recently learned, I just need to figure out where to place the code for the culling. I'm guessing in my method that draws all my cubes but I could be wrong. My camera class currently implements a bounding frustum which is in the update method like so frustum.Matrix = (view * proj); Simple enough, as I can call that when I have a camera object in my class. This works for now, as I only have a camera in my main game class. The problem comes when I decide to move my camera to my player class, but I can worry about that later. ContainmentType CurrentContainmentType = ContainmentType.Disjoint; CurrentContainmentType = CamerasFrustrum.Contains(cubes.CollisionSphere); Can it really be as easy as adding those two lines to my foreach loop in my draw method? Or am I missing something bigger here? UPDATE: I have added the lines to my draw methods and it works great!! So great infact that just moving a little bit removes the whole map. Many factors could of caused this, so I'll try to break it down. cubeBoundingSphere = new BoundingSphere(cubePosition, 0.5f); This is in my cube constructor. cubePosition is stored in an array, The vertices that define my cube are factors of 1 ie: (1,0,1) so the radius should be .5. I least I think it should. The spheres are created every time a cube is created of course. ContainmentType CurrentContainmentType = ContainmentType.Disjoint; foreach (Cube block in cube.cubes) { CurrentContainmentType = cam.frustum.Contains(cube.cubeBoundingSphere); ///more code here if (CurrentContainmentType != ContainmentType.Disjoint) { cube.Draw(effect); } Within my draw method. Now I know this works because the map disappears, its just working wrong. Any idea on what I'm doing wrong?

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• What does path finding in internet routing do and how is it different from A*?

  - by alan2here

  Note: If you don't understand this question then feel free to ask clarification in the comments instead of voting down, it might be that this question needs some more work at the moment. I've been directed here from the Stack Excange chat room Root Access because my question didn't fit on Super User. In many aspects path finding algorithms like A star are very similar to internet routing. For example: A node in an A* path finding system can search for a path though edges between other nodes. A router that's part of the internet can search for a route though cables between other routers. In the case of A*, open and closed lists are kept by the system as a whole, sepratly from any individual node as well as each node being able to temporarily store a state involving several numbers. Routers on the internet seem to have remarkable properties, as I understand it: They are very performant. New nodes can be added at any time that use a free address from a finite (not tree like) address space. It's real routing, like A*, there's never any doubling back for example. Similar IP addresses don't have to be geographically nearby. The network reacts quickly to changes to the networks shape, for example if a line is down. Routers share information and it takes time for new IP's to be registered everywhere, but presumably every router doesn't have to store a list of all the addresses each of it's directions leads most directly to. I'm looking for a basic, general, high level description of the algorithms workings from the point of view of an individual router. Does anyone have one? I presume public internet routers don't use A* as the overheads would be to large, and scale to poorly. I also presume there is a single method worldwide because it seems as if must involve a lot of transferring data to update and communicate a reasonable amount of state between neighboring routers. For example, perhaps the amount of data that needs to be stored in each router scales logarithmically with the number of routers that exist worldwide, the detail and reliability of the routing is reduced over increasing distances, there is increasing backtracking involved in parts of the network that are less geographically uniform or maybe each router really does perform an A* style search, temporarily maintaining open and closed lists when a packet arrives.

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• Are closures with side-effects considered "functional style"?

  - by Giorgio

  Many modern programming languages support some concept of closure, i.e. of a piece of code (a block or a function) that Can be treated as a value, and therefore stored in a variable, passed around to different parts of the code, be defined in one part of a program and invoked in a totally different part of the same program. Can capture variables from the context in which it is defined, and access them when it is later invoked (possibly in a totally different context). Here is an example of a closure written in Scala: def filterList(xs: List[Int], lowerBound: Int): List[Int] = xs.filter(x => x >= lowerBound) The function literal x => x >= lowerBound contains the free variable lowerBound, which is closed (bound) by the argument of the function filterList that has the same name. The closure is passed to the library method filter, which can invoke it repeatedly as a normal function. I have been reading a lot of questions and answers on this site and, as far as I understand, the term closure is often automatically associated with functional programming and functional programming style. The definition of function programming on wikipedia reads: In computer science, functional programming is a programming paradigm that treats computation as the evaluation of mathematical functions and avoids state and mutable data. It emphasizes the application of functions, in contrast to the imperative programming style, which emphasizes changes in state. and further on [...] in functional code, the output value of a function depends only on the arguments that are input to the function [...]. Eliminating side effects can make it much easier to understand and predict the behavior of a program, which is one of the key motivations for the development of functional programming. On the other hand, many closure constructs provided by programming languages allow a closure to capture non-local variables and change them when the closure is invoked, thus producing a side effect on the environment in which they were defined. In this case, closures implement the first idea of functional programming (functions are first-class entities that can be moved around like other values) but neglect the second idea (avoiding side-effects). Is this use of closures with side effects considered functional style or are closures considered a more general construct that can be used both for a functional and a non-functional programming style? Is there any literature on this topic? IMPORTANT NOTE I am not questioning the usefulness of side-effects or of having closures with side effects. Also, I am not interested in a discussion about the advantages / disadvantages of closures with or without side effects. I am only interested to know if using such closures is still considered functional style by the proponent of functional programming or if, on the contrary, their use is discouraged when using a functional style.

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• If most of team can't follow the architecture, what do you do?

  - by Chris

  Hi all, I'm working on a greenfields project with two other developers. We're all contractors, and myself and one other just started working on the project while the orginal one has been doing most of the basic framework coding. In the past month, my fellow programmer and I have been just frustrated by the design descisions done by our co-worker. Here's a little background information: The application at face value appeared to be your standard n-layered web application using C# on the 3.5 framework. We have a data layer, business layer and a web interface. But as we got deeper into the project we found some very interesting things that have caused us some troubles. There is a custom data access sqlHelper type base which only accepts dictionary key/valued entries and returns only data tables. There are no entity objects, but there are some massive objects which do everything and then are tossed into session for persitance. The general idea is that the pages (.aspx) don't do anything, while the controls (.ascx) do everything. The general flow is that a client clicks on a button, which goes to a user control base which passes a process request to the 'BLL' class which goes to the page processor, which then goes to a getControlProcessor, which at last actually processes the request. The request itself is made up of a dictionary which is passing a string valued method name, stored procedure name, a control name and possibly a value. All switching of the processing is done by comparing the string values of the control names and method names. Pages are linked together via a common header control that uses a combination of javascript and tables to create a hyperlink effect. And as I found out yesterday, a simple hyperlink between one page and another does not work because of the need to have quite a bit of information in session to determine which control to display on a page. My fellow programmer and I both believe that this is a strange and uncommon approach to web application development. Both of us have been in this business for over five years and neither of us have seen this approach. My question is this, how would we approach our co-worker and voice our concerns and what should we do if he does not want to accept the criteic? We both do not want to insult the work that has been done, but feel that going forward will create a nightmare for development. Thanks for your comments.

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• Adding JavaScript to your code dependent upon conditions

  - by DavidMadden

  You might be in an environment where you code is source controlled and where you might have build options to different environments.  I recently encountered this where the same code, built on different configurations, would have the website at a different URL.  If you are working with ASP.NET as I am you will have to do something a bit crazy but worth while.  If someone has a more efficient solution please share. Here is what I came up with to make sure the client side script was placed into the HEAD element for the Google Analytics script.  GA wants to be the last in the HEAD element so if you are doing others in the Page_Load then you should do theirs last. The settings object below is an instance of a class that holds information I collection.  You could read from different sources depending on where you stored your unique ID for Google Analytics. *** This has been formatted to fit this screen. *** if (!IsPostBack) { if (settings.GoogleAnalyticsID != null || settings.GoogleAnalyticsID != string.Empty) { string str = @"//<!CDATA[ var _gaq = _gaq || []; _gaq.push(['_setAccount', '"  + settings.GoogleAnalyticsID + "']); _gaq.push(['_trackPageview']);  (function () {  var ga = document.createElement('script');  ga.type = 'text/javascript';  ga.async = true;  ga.src = ('https:' == document.location.protocol  ? 'https://ssl' :  'http://www') + '.google-analytics.com/ga.js'; var s = document.getElementsByTagName('script')[0];  s.parentNode.insertBefore(ga, s);})();"; System.Web.UI.HtmlControls.HtmlGenericControl si =  new System.Web.UI.HtmlControls.HtmlGenericControl(); si.TagName = "script"; si.Attributes.Add("type", @"text/javascript"); si.InnerHtml = sb.ToString(); this.Page.Header.Controls.Add(si); } } The code above will prevent the code from executing if it is a PostBack and then if the ID was not able to be read or something caused the settings to be lost by accident. If you have larger function to declare, you can use a StringBuilder to separate the lines. This is the most compact I wished to go and manage readability.

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• linux raid 1: right after replacing and syncing one drive, the other disk fails - understanding what is going on with mdstat/mdadm

  - by devicerandom

  We have an old RAID 1 Linux server (Ubuntu Lucid 10.04), with four partitions. A few days ago /dev/sdb failed, and today we noticed /dev/sda had pre-failure ominous SMART signs (~4000 reallocated sector count). We replaced /dev/sdb this morning and rebuilt the RAID on the new drive, following this guide: http://www.howtoforge.com/replacing_hard_disks_in_a_raid1_array Everything went smooth until the very end. When it looked like it was finishing to synchronize the last partition, the other old one failed. At this point I am very unsure of the state of the system. Everything seems working and the files seem to be all accessible, just as if it synchronized everything, but I'm new to RAID and I'm worried about what is going on. The /proc/mdstat output is: Personalities : [raid1] [linear] [multipath] [raid0] [raid6] [raid5] [raid4] [raid10] md3 : active raid1 sdb4[2](S) sda4[0] 478713792 blocks [2/1] [U_] md2 : active raid1 sdb3[1] sda3[2](F) 244140992 blocks [2/1] [_U] md1 : active raid1 sdb2[1] sda2[2](F) 244140992 blocks [2/1] [_U] md0 : active raid1 sdb1[1] sda1[2](F) 9764800 blocks [2/1] [_U] unused devices: <none> The order of [_U] vs [U_]. Why aren't they consistent along all the array? Is the first U /dev/sda or /dev/sdb? (I tried looking on the web for this trivial information but I found no explicit indication) If I read correctly for md0, [_U] should be /dev/sda1 (down) and /dev/sdb1 (up). But if /dev/sda has failed, how can it be the opposite for md3 ? I understand /dev/sdb4 is now spare because probably it failed to synchronize it 100%, but why does it show /dev/sda4 as up? Shouldn't it be [__]? Or [_U] anyway? The /dev/sda drive now cannot even be accessed by SMART anymore apparently, so I wouldn't expect it to be up. What is wrong with my interpretation of the output? I attach also the outputs of mdadm --detail for the four partitions: /dev/md0: Version : 00.90 Creation Time : Fri Jan 21 18:43:07 2011 Raid Level : raid1 Array Size : 9764800 (9.31 GiB 10.00 GB) Used Dev Size : 9764800 (9.31 GiB 10.00 GB) Raid Devices : 2 Total Devices : 2 Preferred Minor : 0 Persistence : Superblock is persistent Update Time : Tue Nov 5 17:27:33 2013 State : clean, degraded Active Devices : 1 Working Devices : 1 Failed Devices : 1 Spare Devices : 0 UUID : a3b4dbbd:859bf7f2:bde36644:fcef85e2 Events : 0.7704 Number Major Minor RaidDevice State 0 0 0 0 removed 1 8 17 1 active sync /dev/sdb1 2 8 1 - faulty spare /dev/sda1 /dev/md1: Version : 00.90 Creation Time : Fri Jan 21 18:43:15 2011 Raid Level : raid1 Array Size : 244140992 (232.83 GiB 250.00 GB) Used Dev Size : 244140992 (232.83 GiB 250.00 GB) Raid Devices : 2 Total Devices : 2 Preferred Minor : 1 Persistence : Superblock is persistent Update Time : Tue Nov 5 17:39:06 2013 State : clean, degraded Active Devices : 1 Working Devices : 1 Failed Devices : 1 Spare Devices : 0 UUID : 8bcd5765:90dc93d5:cc70849c:224ced45 Events : 0.1508280 Number Major Minor RaidDevice State 0 0 0 0 removed 1 8 18 1 active sync /dev/sdb2 2 8 2 - faulty spare /dev/sda2 /dev/md2: Version : 00.90 Creation Time : Fri Jan 21 18:43:19 2011 Raid Level : raid1 Array Size : 244140992 (232.83 GiB 250.00 GB) Used Dev Size : 244140992 (232.83 GiB 250.00 GB) Raid Devices : 2 Total Devices : 2 Preferred Minor : 2 Persistence : Superblock is persistent Update Time : Tue Nov 5 17:46:44 2013 State : clean, degraded Active Devices : 1 Working Devices : 1 Failed Devices : 1 Spare Devices : 0 UUID : 2885668b:881cafed:b8275ae8:16bc7171 Events : 0.2289636 Number Major Minor RaidDevice State 0 0 0 0 removed 1 8 19 1 active sync /dev/sdb3 2 8 3 - faulty spare /dev/sda3 /dev/md3: Version : 00.90 Creation Time : Fri Jan 21 18:43:22 2011 Raid Level : raid1 Array Size : 478713792 (456.54 GiB 490.20 GB) Used Dev Size : 478713792 (456.54 GiB 490.20 GB) Raid Devices : 2 Total Devices : 2 Preferred Minor : 3 Persistence : Superblock is persistent Update Time : Tue Nov 5 17:19:20 2013 State : clean, degraded Active Devices : 1 Working Devices : 2 Failed Devices : 0 Spare Devices : 1 Number Major Minor RaidDevice State 0 8 4 0 active sync /dev/sda4 1 0 0 1 removed 2 8 20 - spare /dev/sdb4 The active sync on /dev/sda4 baffles me. I am worried because if tomorrow morning I have to replace /dev/sda, I want to be sure what should I sync with what and what is going on. I am also quite baffled by the fact /dev/sda decided to fail exactly when the raid finished resyncing. I'd like to understand what is really happening. Thanks a lot for your patience and help. Massimo

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• Separating logic and data in browser game

  - by Tesserex

  I've been thinking this over for days and I'm still not sure what to do. I'm trying to refactor a combat system in PHP (...sorry.) Here's what exists so far: There are two (so far) types of entities that can participate in combat. Let's just call them players and NPCs. Their data is already written pretty well. When involved in combat, these entities are wrapped with another object in the DB called a Combatant, which gives them information about the particular fight. They can be involved in multiple combats at once. I'm trying to write the logic engine for combat by having combatants injected into it. I want to be able to mock everything for testing. In order to separate logic and data, I want to have two interfaces / base classes, one being ICombatantData and the other ICombatantLogic. The two implementers of data will be one for the real objects stored in the database, and the other for my mock objects. I'm now running into uncertainties with designing the logic side of things. I can have one implementer for each of players and NPCs, but then I have an issue. A combatant needs to be able to return the entity that it wraps. Should this getter method be part of logic or data? I feel strongly that it should be in data, because the logic part is used for executing combat, and won't be available if someone is just looking up information about an upcoming fight. But the data classes only separate mock from DB, not player from NPC. If I try having two child classes of the DB data implementer, one for each entity type, then how do I architect that while keeping my mocks in the loop? Do I need some third interface like IEntityProvider that I inject into the data classes? Also with some of the ideas I've been considering, I feel like I'll have to put checks in place to make sure you don't mismatch things, like making the logic for an NPC accidentally wrap the data for a player. Does that make any sense? Is that a situation that would even be possible if the architecture is correct, or would the right design prohibit that completely so I don't need to check for it? If someone could help me just layout a class diagram or something for this it would help me a lot. Thanks. edit Also useful to note, the mock data class doesn't really need the Entity, since I'll just be specifying all the parameters like combat stats directly instead. So maybe that will affect the correct design.

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• Is this an effective monetization method for an Android game? [on hold]

  - by Matthew Page

  The short version: I plan to make an Android puzzle game where the user tries to get 3-6 numbers to their predetermined goal numbers. The free version of the app will have three predetermined levels (easy, medium, hard). The full version ($0.99, probably) will have a level generator where there will be unlimited easy, medium, or hard levels, as well as a custom difficulty option where users can set specific vales to the number of numbers to equate to their goal, the number of buttons to use, etc. Users will also have the option to get a one-time "hint" for a fee of $0.49, or unlimited hints for a one-time fee of $2.99. The long version: Mechanics of Game and Victory The application is a number puzzle. When the user begins a new game, depending on the input by the user, between 3 and 6 numbers show up on the top of the screen, and between 3 and 6 buttons show up on the bottom of the screen. The buttons all have two options: to increase every number the same way, or decrease every number the same way. The buttons either use addition / subtraction, multiplication / division, or exponents / roots, all depending on the number displayed on the button. Addition buttons are green, multiplication buttons are blue, and exponential buttons are red. The user wins when all of the numbers displayed on the screen equate to their goal number, displayed below each number. Monetization If the user is playing the full (priced) version of the app, upon the start of the game, the user will be confronted with a dialogue asking for the number of buttons and the number of numbers to equate in the game. Then, based on the user input, a random puzzle will be generated. If the user is playing the free version of the app, the user will be asked to either play an “easy”, “hard”, or “expert” puzzle. A pre-determined puzzle from each category will be used in the game. If the user has played that puzzle before, a dialogue will show saying this to the user and advertising the full version of the app. The full version of the app will also be advertised upon the successful or in successful completion of a puzzle. Upon exiting this advertisement, another full screen advertisement will appear from a third party. Also, the solution to the puzzle should be stored by the program, and if the user pays a small fee, he/she can see a hint to the solution to the program. In the free version of the app, the user may use their first hint for free. Also, the user can use unlimited hints for a slightly larger fee. Is this an effective monetization method?

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• What is the most appropriate testing method in this scenario?

  - by Daniel Bruce

  I'm writing some Objective-C apps (for OS X/iOS) and I'm currently implementing a service to be shared across them. The service is intended to be fairly self-contained. For the current functionality I'm envisioning there will be only one method that clients will call to do a fairly complicated series of steps both using private methods on the class, and passing data through a bunch of "data mangling classes" to arrive at an end result. The gist of the code is to fetch a log of changes, stored in a service-internal data store, that has occurred since a particular time, simplify the log to only include the last applicable change for each object, attach the serialized values for the affected objects and return this all to the client. My question then is, how do I unit-test this entry point method? Obviously, each class would have thorough unit tests to ensure that their functionality works as expected, but the entry point seems harder to "disconnect" from the rest of the world. I would rather not send in each of these internal classes IoC-style, because they're small and are only made classes to satisfy the single-responsibility principle. I see a couple possibilities: Create a "private" interface header for the tests with methods that call the internal classes and test each of these methods separately. Then, to test the entry point, make a partial mock of the service class with these private methods mocked out and just test that the methods are called with the right arguments. Write a series of fatter tests for the entry point without mocking out anything, testing the entire functionality in one go. This looks, to me, more like "integration testing" and seems brittle, but it does satisfy the "only test via the public interface" principle. Write a factory that returns these internal services and take that in the initializer, then write a factory that returns mocked versions of them to use in tests. This has the downside of making the construction of the service annoying, and leaks internal details to the client. Write a "private" initializer that take these services as extra parameters, use that to provide mocked services, and have the public initializer back-end to this one. This would ensure that the client code still sees the easy/pretty initializer and no internals are leaked. I'm sure there's more ways to solve this problem that I haven't thought of yet, but my question is: what's the most appropriate approach according to unit testing best practices? Especially considering I would prefer to write this test-first, meaning I should preferably only create these services as the code indicates a need for them.

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• rotating an object on an arc

  - by gardian06

  I am trying to get a turret to rotate on an arc, and have hit a wall. I have 8 possible starting orientations for the turrets, and want them to rotate on a 90 degree arc. I currently take the starting rotation of the turret, and then from that derive the positive, and negative boundary of the arc. because of engine restrictions (Unity) I have to do all of my tests against a value which is between [0,360], and due to numerical precision issues I can not test against specific values. I would like to write a general test without having to go in, and jury rig cases //my current test is: // member variables public float negBound; public float posBound; // found in Start() function (called immediately after construction) // eulerAngles.y is the the degree measure of the starting y rotation negBound = transform.eulerAngles.y-45; posBound = transform.eulerAngles.y+45; // insure that values are within bounds if(negBound<0){ negBound+=360; }else if(posBound>360){ posBound-=360; } // called from Update() when target not in firing line void Rotate(){ // controlls what direction if(transform.eulerAngles.y>posBound){ dir = -1; } else if(transform.eulerAngles.y < negBound){ dir = 1; } // rotate object } follows is a table of values for my different cases (please excuse my force formatting) read as base is the starting rotation of the turret, neg is the negative boundry, pos is the positive boundry, range is the acceptable range of values, and works is if it performs as expected with the current code. |base-|-neg-|-pos--|----------range-----------|-works-| |---0---|-315-|--45--|-315-0,0-45----------|----------| |--45--|---0---|--90--|-0-45,54-90----------|----x----| |-135-|---90--|-180-|-90-135,135-180---|----x----| |-180-|--135-|-225-|-135-180,180-225-|----x----| |-225-|--180-|-270-|-180-225,225-270-|----x----| |-270-|--225-|-315-|-225-270,270-315-|----------| |-315-|--270-|---0---|--270-315,315-0---|----------| I will need to do all tests from derived, or stored values, but can not figure out how to get all of my cases to work simultaneously. //I attempted to concatenate the 2 tests: if((transform.eulerAngles.y>posBound)&&(transform.eulerAngles.y < negBound)){ dir *= -1; } this caused only the first case to be successful // I attempted to store a opposite value, and do a void Rotate(){ // controlls what direction if((transform.eulerAngles.y > posBound)&&(transform.eulerAngles.y<oposite)){ dir = -1; } else if((transform.eulerAngles.y < negBound)&&(transform.eulerAngles.y>oposite)){ dir = 1; } // rotate object } this causes the opposite situation as indicated on the table. What am I missing here?

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• Hierachies....from the Top Down

  - by Joe G

  I've been struggling with how to write on the topic of the importance of hierarchy design.  It's not so much that hierarchies haven't always been important, it's more of that with Fusion, the timing of when the hierarchies are designed should take a higher priority.    I will attempt to explain..... When I was implementing applications, back in the day, we had the list of detailed account values to enter with the obvious parent accounts. Then, after the setup was complete and things were functioning, the reporting phase started.  Users explained the elements that they want on the reports, what totals should be included, and how things should be compared.  Frequently, there was at least one calculation that became a nightmare either because it was based on very specific things that didn't relate to anything else or because it was "hardcoded" so that when something changed, someone need to "fix" the report. With Fusion, the process changes slightly.  You still want to enter all of the detailed accounts, but before you start adding parent values, you should investigate the reporting requirements from the top-down.  It's better to build hierarchies based on the reporting requirements than it is to build reports based on random hierarchies. Build reports based on hierarchies that resemble the reports themselves, and maintain the hierarchies without rework of the reports. For example, if you look at an income statement, you may have line items for Material Costs, Employee Costs, Travel & Entertainment, and Total Operating Expenses.  In your hierarchy, you have detail values that roll up to Material Costs, Employee Costs, and Travel & Entertainment which roll up to Total Operating Expenses. Balances are stored automatically in the cube for each of these.  When you define the report, you pick each of these members - no calculations required.  If a new detail value is added, you simply add it to the hierarchy, and there is no need to modify the report. I realize that there are always exceptions that require special handling, but I am confident that you will end up with much fewer exceptions if you make reporting a priority and design your hierarchies from the top-down.

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• SDL - Getting a single keypress event instead of a keystate?

  - by MrKatSwordfish

  Right now I'm working on a simple SDL project, but I've hit an issue when trying to get a single keypress event to skip past a splash screen. Right now, there are 4 start-up splash screens that I would like to be able to skip with a single keypress (of any key). My issue is that, as of now, if I hold down a key, it skips through each splash screen to the very last one immediately. The splash screens are stored as an array of SDL surfaces which are all loaded at the initialization of the state. I have an variable called currentSplashImage that controls which element of the array is being rendered on the screen. I've set it up so that whenever there's a SDL_KEYDOWN event, it triggers a single incrementation of the currentSplashImage variable. So, I'm really not sure why my code isn't working correctly. For some reason, when I hold down a button, it seems to be treating the held button as a new key press event every time it ticks through the code. Does anyone know how I can go about fixing this issue? [Here's a snippet of code that I've been using...] void SplashScreenState::handleEvents() { SDL_PollEvent( &localEvent ); if ( localEvent.type == SDL_KEYDOWN ) { if ( currentSplashImage < 3 && currentSplashImage >= 0) { currentSplashImage++; } } else if ( localEvent.type == SDL_QUIT ) { smgaEngine.setRunning(false); } } I should also mention that the SDL_Event 'localEvent' is part of the GameState parent class, while this event handling code is part of a SplashScreenState subclass. If anyone knows why this is happening, or if there is any way to improve my code, It'd be helpful to me! :D I'm still a very new programmer, trying to learn. UPDATE: I added a std::cout line to that the code runs multiple times with a single KEYDOWN event. I also tried disabling SDL_EnableKeyRepeat, but it didn't fix the issue. void SplashScreenState::handleEvents() { SDL_PollEvent( &localEvent ); if ( localEvent.type == SDL_KEYDOWN ) { if ( currentSplashImage < 3 && currentSplashImage >= 0) { currentSplashImage++; std::cout << "KEYDOWN.."; //<---- test cout line } } else if ( localEvent.type == SDL_QUIT ) { smgaEngine.setRunning(false); } } This prints out "KEYDOWN..KEYDOWN..KEYDOWN.." in the cout stream when a button is held.

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• How is a switch statement better than a series of if statements? [closed]

  - by user1276078

  Possible Duplicate: Should I use switch statements or long if…else chains? I'm working on a small program that will conduct an Insertion Sort. A number will be inputted through the keyboard and stored in a variable I called "num." I've decided to use a switch statement in order to obtain the number inputted. switch( e.getKeyCode() ) { case KeyEvent.VK_0: num = 0; break; case KeyEvent.VK_1: num = 1; break; case KeyEvent.VK_2: num = 2; break; case KeyEvent.VK_3: num = 3; break; case KeyEvent.VK_4: num = 4; break; case KeyEvent.VK_5: num = 5; break; case KeyEvent.VK_6: num = 6; break; case KeyEvent.VK_7: num = 7; break; case KeyEvent.VK_8: num = 8; break; case KeyEvent.VK_9: num = 9; break; } I realized one other course of action could have been to use a set of if statements. if( e.getKeyCode() == KeyEvent.VK_0 ) num = 0; else if( e.getKeyCode() == KeyEvent.VK_1 ) num = 1; etc. for every number up until 9. I then wondered what the essential difference is between a switch statement and a series of if statements. I know it saves space and time to write, but it's not that much. So, my question is, aside from the space, does a switch statement differ from a series of if statments in any way? Is it faster, less error-prone, etc.? This question really doesn't affect my code that much. I was just wondering. Also, this question pertains to the JAVA language, not any other programming language.

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• Apache Getting Bogged Down By Certain Script (Wp-Cron.php) - How To Kill Process Automatically

  - by user50037

  I have a server that is running a number of wordpress blogs, and a number of them have several hundred/thousand posts. Every couple of days, the server slows to a crawl due to a file being run on Wordpress called WP-cron.php. My entire apache process log turns into this : http:// imgur.com/A7K9k.png Times that by quite a bit. And server no go. Each process takes up about 1.1% of ram. And when we have 50 of them on the go. It gets insane. Not all of them are coming from the same blog, they are pretty widespread. In the Apache process page of WHM, they are usually ALL set to the status of "C", which means closing. But they can sit there until they crash the server, and they still hold the memory. Just google "wp-cron.php load" and you will find plenty of people with similar issues. In anycase, we have think it is down to users adding a tonne of dead "pinglists" to their wordpress installation. Which in turn wordpress loops through them endlessly. Problem number 1. Does anyone have any other suggestions about what would cause the Wordpress file wp-cron.php to loop endlessly. I still think it is down to pings, because all of the people we have contacted about their account load going sky high, have had massive ping lists. Problem number 2. Even if it is down to excessive pinglists in wordpress. We cannot be babying every single account on the server waiting for it to start spawning the wp-cron processes. It often happens overnight, and I start getting SMS alerts at 2am about the load. I have CSF installed, which apparently would have ended the processes if they ran over XXX time. But I have been told that it won't catch the processes because they end up in this state of "closing" (They show up as "C" on the Apache page of WHM). Apparently CSF will only kill processes that are "running" which C does not count. I have seen various other scripts such as : http://dltj.org/article/die-apache-die/ . I took a look at the stat of /proc. But I was boggled at which delimited part was the time running. And if there was any way I could connect it back to an actual Apache process, so that I could see what file was running (So only close connections connected to wp-cron.php, with a state of "C"). Overall I know Problem 2 glosses over the real reason. But I do put the whole thing to excessive pinglists in Wordpress. But I just cannot sit there and babysit every single installation 24/7. So I need a way to save the server when I am not available. Any help would be much appreciated.

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• Backup, Migrate or Clone Failing CentOS 4 (LVM)

  - by Hegelworm

  I've been running a BlueQuartz CentOS 4 system (Nuonce.net distro) for a few years now and although the hard drive (Deskstar) has always been a bit noisy, on a few recent occasions I've heard it having trouble spinning up. Basically, I want to clone this drive to a similar sized one (80 Gig). I've spent many hours reading upon dd, dd_rescue, rsync, clonezilla and LVM mirroring yet the sheer number of options and nightmarish accounts has left me frozen - unable to make an informed decision as to how to start. I've made a few attempts. dd failed after about 2 hours, as, although the drives appeared to be identical on the surface (ATA Seagate Barracudas, Thai not Chinese), the destination drive is slightly smaller. My most recent attempt involved using a Debian CD to format the new drive and then rsync-ing everything over and editing the new drive's grub and fstab to reflect the changes. No joy here either as I hadn't chosen LVM when partitioning the destination drive and it wouldn't boot. As you can probably tell, I'm out of my depth here and a panic-invoking mixture of caution and frustration has prompted me to sign up here. The server itself, although not strictly a production environment, has a very specific installation of Festival, LAME and ffMpeg and provides the back-end for a Text-to-Speech jQuery plugin that I've built over the last 2 years. I'm also planning to rebuild the whole TTS system on Debian as the existing CentOS system still has PHP4 etc. For now though, I'd really like to just shift everything over to a new drive. As this is my first post, please feel free to lay any house rules on me that I might've overlooked; I've been hovering around StackOverflow for a while now but have only just signed up. Many thanks. Update: Thanks for your responses so far - it's much appreciated and makes me feel a little more confident when I can double-check things here. I had the idea of doing a fresh install of CentOS (from the original disk) on the new drive so the partitions and LVM were all set up correctly (after disconnecting my source drive to prevent painful mistakes). I then booted into rescue mode from the same CD, and, to avoid a conflicting label, changed the /boot partition's label using e2label to /bootnew. I then changed the VolGroup name using lvm vgrename from VolGroup00 to VolGroup001. I could then boot with both drives in. After mounting the new drive (via its VolGroup001 alias) into /newhd, I rsync-ed over everything I could to the new drive, using -avr switches and backslashes. Like mentioned here. I then disconnected my original source drive again, booted from the liveCD again, changed back the boot partition label from /bootnew to /boot using e2label and then renamed the VolGroup back to VolGroup00. I then rebooted and it went through the familiar start-up routine only to not find a host of files in proc, usr, lib, var etc. The boot did complete but there were lots of red 'FAILS'. I could log in with my existing creds, but the network was kaput, I couldn't startX (desktop GUI) and there were also a few (a lot) of error messages pertaining to iptables. Back to square one. I naively thought I'd nailed it. Shall I just buy a bigger hard drive and attempt the dd route? I've read that this can mess with LVM setups and there's the added risk of working on two unmounted drives at once with a low-level tool. Thanks again.

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• How to synchronize the ball in a network pong game?

  - by Thaars

  I’m developing a multiplayer network pong game, my first game ever. The current state is, I’ve running the physic engine with the same configurations on the server and the clients. The own paddle movement is predicted and get just confirmed by the authoritative server. Is a difference detected between them, I correct the position at the client by interpolation. The opponent paddle is also interpolated 200ms to 100ms in the past, because the server is broadcasting snapshots every 100ms to each client. So far it works very well, but now I have to simulate the ball and have a problem to understanding the procedure. I’ve read Valve’s (and many other) articles about fast-paced multiplayer several times and understood their approach. Maybe I can compare my ball with their bullets, but their advantage is, the bullets are not visible. When I have to display the ball, and see my paddle in the present, the opponent in the past and the server is somewhere between it, how can I synchronize the ball over all instances and ensure, that it got ever hit by the paddle even if the paddle is fast moving? Currently my ball’s position is simply set by a server update, so it can happen, that the ball bounces back, even if the paddle is some pixel away (because of a delayed server position). Until now I’ve got no synced clock over all instances. I’m sending a client step index with each update to the server. If the server did his job, he sends the snapshot with the last step index of each client back to the clients. Now I’m looking for the stored position at the returned step index and compare them. Do I need a common clock to sync the ball? EDIT: I've tried to sync a common clock for the server and all clients with a timestamp. But I think it's better to use an own stepping instead of a timestamp (so I don't need to calculate with the ping and so on - and the timestamp will never be exact). The physics are running 60 times per second and now I use this for keeping them synchronized. Is that a good way? When the ball gets calculated by each client, the angle after bouncing can differ because of the different position of the paddles (the opponent is 200ms in the past). When the server is sending his ball position, velocity and angle (because he knows the position of each paddle and is authoritative), the ball could be in a very different position because of the different angles after bouncing (because the clients receive the server data after 100ms). How is it possible to interpolate such a huge difference? I posted this question some days ago at stackoverflow, but got no answer yet. Maybe this is the better place for this question.

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• Ad-hoc reporting similar to Microstrategy/Pentaho - is OLAP really the only choice (is OLAP even sufficient)?

  - by TheBeefMightBeTough

  So I'm getting ready to develop an API in Java that will provide all dimensions, metrics, hierarchies, etc to a user such that they can pick and choose what they want (say, e.g., dimensions of Location (a store) and Weekly, and the metric Product Sales $), provide their choices to the api, and have it spit out an object that contains the answer to their question (the object would probably be a set of cells). I don't even believe there will be much drill up/down. The data warehouse the APIwill interface with is in a standard form (FACT tables, dimensions, star schema format). My question is, is an OLAP framework such as Mondrian the only way to achieve something akin to ad-hoc reporting? I can envisage a really large Cube (or VirtualCube) that contains most of the dimensions and metrics the user could ever want, which would give the illusion of ad-hoc reporting. The problem is that there is a ton of setup to do (so much XML) to get the framework to work with the data. Further it requires specific knowledge, such as MDX, and even moreso learning the framework peculiars (Mondrian API). Finally, I am not positive it will scale much better than simply making queries against a SQL database. OLAP to me feels like very old technology. Is performance really an issue anymore? The alternative I can think of would be dynamic SQL. If the existing tables in the data warehouse conform to a naming scheme (FACT_, DIM_, etc), or if a very simple config file/ database table containing config information existed that stored which tables are fact tables, which are dimensions, and what metrics are available, then couldn't the api read from that and assembly the appropriate sql query? Would this necessarily be harder than learning MDX, Mondrian (or another OLAP framework), and creating all the cubes? In general, I feel that OLAP is at the same time too powerful (supports drill up/down, complex functions) and outdated and am reluctant to base my architecture on it. However, I am unsure if the alternative(s), such as rolling my own ad-hoc reporting framework using dynamic SQL would remove any complexity while still fulfilling requirements, both functional and non-functional (e.g., scalability; some FACT tables have many millions of rows). I also wonder about other techniques (e.g., hive). Has anyone here tried to do ad-hoc reporting? Any advice? I expect this project to take a pretty long time (3 months min, but probably longer), so I just do not want to commit to an architecture without being absolutely sure of its pros and cons. Thanks so much.

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• SQL Server Log File Won't Shrink due cause "log are pending replication" on non replicated DB?

  - by user796466

  I have a non Mission Critial DB 9am-5pm SQL Server database that I have set up to do nightly full backups and log backups every 30 minutes during business hours. The database is in full recovery and normally I have no reason to truncate/shrink logs unless I do some heavy maintenance. Log backups manage the size with no issue. However I have not been at this client for several weeks and upon inspection I noticed that the log had grown to about 10 times the size of the .mdf file. I poked around backups had been running and I had not gotten any severity error alerts (SQL mail). I attempted to put DB in simple recovery and shrink the log, this was no good. I precede to try a log backup and I got: The log was not truncated because records at the beginning of the log are pending replication or Change Data Capture. Ensure the Log Reader Agent or capture job is running or use sp_repldone to mark transactions as distributed or captured. Restart SQL Server rinse repeat same thing ... I said ??? Replication is not nor ever has been set up on this DB or database /server ??? So the log backups have not been flushing the .ldf. So I did a couple hours of research and I found: http://www.sqlmonster.com/Uwe/Forum.aspx/sql-server/5445/Log-file-is-not-truncated-inspite-of-regular-log-backup http://www.eggheadcafe.com/software/aspnet/30708322/the-log-was-not-truncated-because-records-at-the-beginning-of-the-log-are-pending-replication.aspx seems to be some kind of poorly documented bug ?? The solution seems to have been to run exec sp_repldone, more precisley EXEC sp_repldone @xactid = NULL, @xact_segno = NULL, @numtrans = 0, @time= 0, @reset = 1 This procedure can be used in emergency situations to allow truncation of the transaction log when transactions pending replication are present. Using this procedure prevents Microsoft SQL Server 2000 from replicating the database until the database is unpublished and republished. ~ MSDN When I do that I get the following Msg 18757, Level 16, State 1, Procedure sp_repldone, Line 1 Unable to execute procedure. The database is not published. Execute the procedure in a database that is published for replication. Which makes sense Because the DB has never been published for replication. I have several questions: A) First and foremost is, WTF is going on ? What is causeing this, I am interested in knowing the why here ? Is this genuinley a bug or is there some aspect of the backup that is not functioning properly that cause's the DB to mimick a replicated state ? Someone please edify me on this. B) Second ... Do I really have to publish / replicate this DB to exec this SP to fix this ??? Sounds crazy or is there some T-SQL that I can put it in a published state exec the proc and be on my way ... C) Third, if I do indeed have to publish this database to exec the SP to release this unneeded mis replicated/intended log , to get my .ldf file and backup back on track. How do I publish the database without an online host that it is asking for ??? I don't generally do this kind of database administration and need some guidance. Sorry if this is too verbose but just voicing the question helps me clarify it ... Thank you in advance for your help

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• openGL textures in bitmap mode

  - by evenex_code

  For reasons detailed here I need to texture a quad using a bitmap (as in, 1 bit per pixel, not an 8-bit pixmap). Right now I have a bitmap stored in an on-device buffer, and am mounting it like so: glBindBuffer(GL_PIXEL_UNPACK_BUFFER, BFR.G[(T+1)%2]); glTexImage2D(GL_TEXTURE_2D, 0, GL_RGB, W, H, 0, GL_COLOR_INDEX, GL_BITMAP, 0); The OpenGL spec has this to say about glTexImage2D: "If type is GL_BITMAP, the data is considered as a string of unsigned bytes (and format must be GL_COLOR_INDEX). Each data byte is treated as eight 1-bit elements..." Judging by the spec, each bit in my buffer should correspond to a single pixel. However, the following experiments show that, for whatever reason, it doesn't work as advertised: 1) When I build my texture, I write to the buffer in 32-bit chunks. From the wording of the spec, it is reasonable to assume that writing 0x00000001 for each value would result in a texture with 1-px-wide vertical bars with 31-wide spaces between them. However, it appears blank. 2) Next, I write with 0x000000FF. By my apparently flawed understanding of the bitmap mode, I would expect that this should produce 8-wide bars with 24-wide spaces between them. Instead, it produces a white 1-px-wide bar. 3) 0x55555555 = 1010101010101010101010101010101, therefore writing this value ought to create 1-wide vertical stripes with 1 pixel spacing. However, it creates a solid gray color. 4) Using my original 8-bit pixmap in GL_BITMAP mode produces the correct animation. I have reached the conclusion that, even in GL_BITMAP mode, the texturer is still interpreting 8-bits as 1 element, despite what the spec seems to suggest. The fact that I can generate a gray color (while I was expecting that I was working in two-tone), as well as the fact that my original 8-bit pixmap generates the correct picture, support this conclusion. Questions: 1) Am I missing some kind of prerequisite call (perhaps for setting a stride length or pack alignment or something) that will signal to the texturer to treat each byte as 8-elements, as it suggests in the spec? 2) Or does it simply not work because modern hardware does not support it? (I have read that GL_BITMAP mode was deprecated in 3.3, I am however forcing a 3.0 context.) 3) Am I better off unpacking the bitmap into a pixmap using a shader? This is a far more roundabout solution than I was hoping for but I suppose there is no such thing as a free lunch.

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• Integrating with a payment provider; Proper and robust OOP approach

  - by ExternalUse

  History We are currently using a so called redirect model for our online payments (where you send the payer to a payment gateway, where he inputs his payment details - the gateway will then return him to a success/failure callback page). That's easy and straight-forward, but unfortunately quite inconvenient and at times confusing for our customers (leaving the site, changing their credit card details with an additional login on another site etc). Intention & Problem description We are now intending to switch to an integrated approach using an exchange of XML requests and responses. My problem is on how to cater with all (or rather most) of the things that may happen during processing - bearing in mind that normally simplicity is robust whereas complexity is fragile. Examples User abort: The user inputs Credit Card details and hits submit. An XML message to the provider's gateway is sent and waiting for response. The user hits "stop" in his browser or closes the window. ignore_user_abort() in PHP may be an option - but is that reliable? might it be better to redirect the user to a "please wait"-page, that in turn opens an AJAX or other request to the actual processor that does not rely on the connection? Database goes away sounds over-complicated, but with e.g. a webserver in the States and a DB in the UK, it has happened and will happen again: User clicks together his order, payment request has been sent to the provider but the response cannot be stored in the database. What approach could I use, using PHP to sort of start an SQL like "Transaction" that only at the very end gets committed or rolled back, depending on the individual steps? Should then neither commit or roll back have happened, I could sort of "lock" the user to prevent him from paying again or to improperly account for payments - but how? And what else do I need to consider technically? None of the integration examples of e.g. Worldpay, Realex or SagePay offer any insight, and neither Google or my search terms were good enough to find somebody else's thoughts on this. Thank you very much for any insight on how you would approach this!

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• Is it unusual for a small company (15 developers) not to use managed source/version control?

  - by LordScree

  It's not really a technical question, but there are several other questions here about source control and best practice. The company I work for (which will remain anonymous) uses a network share to host its source code and released code. It's the responsibility of the developer or manager to manually move source code to the correct folder depending on whether it's been released and what version it is and stuff. We have various spreadsheets dotted around where we record file names and versions and what's changed, and some teams also put details of different versions at the top of each file. Each team (2-3 teams) seems to do this differently within the company. As you can imagine, it's an organised mess - organised, because the "right people" know where their stuff is, but a mess because it's all different and it relies on people remembering what to do at any one time. One good thing is that everything is backed up on a nightly basis and kept indefinitely, so if mistakes are made, snapshots can be recovered. I've been trying to push for some kind of managed source control for a while, but I can't seem to get enough support for it within the company. My main arguments are: We're currently vulnerable; at any point someone could forget to do one of the many release actions we have to do, which could mean whole versions are not stored correctly. It could take hours or even days to piece a version back together if necessary We're developing new features along with bug fixes, and often have to delay the release of one or the other because some work has not been completed yet. We also have to force customers to take versions that include new features even if they just want a bug fix, because there's only really one version we're all working on We're experiencing problems with Visual Studio because multiple developers are using the same projects at the same time (not the same files, but it's still causing problems) There are only 15 developers, but we all do stuff differently; wouldn't it be better to have a standard company-wide approach we all have to follow? My questions are: Is it normal for a group of this size not to have source control? I have so far been given only vague reasons for not having source control - what reasons would you suggest could be valid for not implementing source control, given the information above? Are there any more reasons for source control that I could add to my arsenal? I'm asking mainly to get a feel for why I have had so much resistance, so please answer honestly. I'll give the answer to the person I believe has taken the most balanced approach and has answered all three questions. Thanks in advance

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• Is the Observer pattern adequate for this kind of scenario?

  - by Omega

  I'm creating a simple game development framework with Ruby. There is a node system. A node is a game entity, and it has position. It can have children nodes (and one parent node). Children are always drawn relatively to their parent. Nodes have a @position field. Anyone can modify it. When such position is modified, the node must update its children accordingly to properly draw them relatively to it. @position contains a Point instance (a class with x and y properties, plus some other useful methods). I need to know when a node's @position's state changes, so I can tell the node to update its children. This is easy if the programmer does something like this: @node.position = Point.new(300,300) Because it is equivalent to calling this: # Code in the Node class def position=(newValue) @position = newValue update_my_children # <--- I know that the position changed end But, I'm lost when this happens: @node.position.x = 300 The only one that knows that the position changed is the Point instance stored in the @position property of the node. But I need the node to be notified! It was at this point that I considered the Observer pattern. Basically, Point is now observable. When a node's position property is given a new Point instance (through the assignment operator), it will stop observing the previous Point it had (if any), and start observing the new one. When a Point instance gets a state change, all observers (the node owning it) will be notified, so now my node can update its children when the position changes. A problem is when this happens: @someNode.position = @anotherNode.position This means that two nodes are observing the same point. If I change one of the node's position, the other would change as well. To fix this, when a position is assigned, I plan to create a new Point instance, copy the passed argument's x and y, and store my newly created point instead of storing the passed one. Another problem I fear is this: somePoint = @node.position somePoint.x = 500 This would, technically, modify @node's position. I'm not sure if anyone would be expecting that behavior. I'm under the impression that people see Point as some kind of primitive rather than an actual object. Is this approach even reasonable? Reasons I'm feeling skeptical: I've heard that the Observer pattern should be used with, well, many observers. Technically, in this scenario there is only one observer at a time. When assigning a node's position as another's (@someNode.position = @anotherNode.position), where I create a whole new instance rather than storing the passed point, it feels hackish, or even inefficient.

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• Zelda-style top-down RPG. How to store tile and collision data?

  - by Delerat

  I'm looking to build a Zelda: LTTP style top-down RPG. I've read a lot on the subject and am currently going back and forth on a few solutions. I'm using C#, MonoGame, and Tiled. For my tile maps, these are the choices I can see in front of me: Store each tile as its own array. Each one having 3-4 layers, texture/animation, depth, flags, and maybe collision(depending on how I do it). I've read warning about memory issues going this route, and my biggest map will probably be 160x120 tiles. My average map however will be about 40x30. The number of tiles might cut in half if I decide to double my tile size, which is currently 16x16. This is the most appealing approach for me, as I feel like I would know how to save maps, make changes, and separate it into chunks for collision checks. Store the static parts of my tile map in multiple arrays acting as the different layers. Then I would just use entities for anything that wasn't static. All of the other tile data such as collisions, depth, etc., would be stored in their own layers as well I guess? This way just seems messy to me though. Regardless of which one I choose, I'm also unsure how to plan all of that other tile data. I could write a bunch of code that would know which integer represents what tile and it's data, but if I changed a tileset in Tiled and exported it again, all of those integers could potentially change and I'd have to adjust a whole bunch of code. My other issue is about how I could do collision. I want to at least support angled collision that slides you around the corners of objects like LTTP does, if not more oddball shapes as well. So do I: Store collision as a flag for binary collision. Could I get this to support angles? Would it be fine to store collision as an integer and have each number represent a certain angle of collision? Store a list of rectangles or other shapes and do collision that way? Sorry for the large two-part(three-part?) question. I felt like these needed to be asked together as I believe each choice influences the other.

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• database independent coding framework options?

  - by statirasystems

  Background: I have not programmed in a while besides doing VBA and a little VB.NET. So please forgive my language use. I'm green and have a head cold. I am reading all I can now, but I have no programming circles to draw from. The information I am providing is to help guide you to what I am looking for. I am not confident I can ask the question properly. Story: I have four different projects that I am starting. Obviously I won't be working on all at the same time however they each will have similar needs and be inter related. They are as follows: Desktop Environment/System User Interface - basically a product that runs on major computers via mono or .net that unifies the look and functions. In the context of the up coming question it would be able to directly access data of various types. It would work in tandum with my office suite, system manager, and network application framework. Office Suite - technically it would not be a suite since I will be doing it from one interfacel except for the Communications Application. As far as the question, it will need to be able to link to various data sources for storing files and using, manipulating, and presenting information. System Manager - an intellegent system to manage and administer the entire network and all equipment. As far as the question, needs to be able to access data for archiving and and for accessing it's own settings stored in various formats, sql or xml. Network Application Framework - A complete system that can be used for ERP, CRM, CMS, Errata, File Management, and so on. As to the question to be able to access it's own or interlink with existing applications. Requirement: C#, Simplifies and reduces coding, use the same code to access diffent databases(ie MySQL, MS SQL, ACCESS, XML, ...), Mono would be nice but not a must, Question: What librarys, frameworks, or other options would be able to help with this? Is there a good resource to guide me? I don't want arguing over what is best, just information to help me further understand and make an educated decision.

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