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  • Release Process Improvements

    - by wallismark
    The process of creating a new build and releasing it to production is a critical step in the SDLC but it is often left as an afterthought and varies greatly from one company to the next. I'm hoping people will share improvements they have made to this process in their organisation so we can all takes steps to 'reduce the pain'. So the question is, specify one painful/time consuming part of your release process and what did you do to improve it? My example: at a previous employer all developers made database changes on one common development database. Then when it came to release time, we used Redgate's SQL Compare to generate a huge script from the differences between the Dev and QA databases. This works reasonably well but the problems with this approach are:- ALL changes in the Dev database are included, some of which may still be 'works in progress'. Sometimes developers made conflicting changes (that were not noticed until the release was in production) It was a time consuming and manual process to create and validate the script (by validate I mean, try to weed out issues like problem 1 and 2). When there were problems with the script (eg the order in which things were run such as creating a record which relies on a foreign key record which is in the script but not yet run) it took time to 'tweak' it so it ran smoothly. It's not an ideal scenario for Continuous Integration. So the solution was:- Enforce a policy of all changes to the database must be scripted. A naming convention was important for ensuring the correct running order of the scripts. Create/Use a tool to run the scripts at release time. Developers had their own copy of the database do develop against (so there was no more 'stepping on each others toes') The next release after we started this process was much faster with fewer problems, indeed the only problems found were due to people 'breaking the rules', eg not creating a script. Once the issues with releasing to QA were fixed, when it came time to release to production it was very smooth. We applied a few other changes (like introducing CI) but this was the most significant, overall we reduced release time from around 3 hours down to a max of 10-15 minutes.

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  • Is the Scala 2.8 collections library a case of "the longest suicide note in history" ?

    - by oxbow_lakes
    First note the inflammatory subject title is a quotation made about the manifesto of a UK political party in the early 1980s. This question is subjective but it is a genuine question, I've made it CW and I'd like some opinions on the matter. Despite whatever my wife and coworkers keep telling me, I don't think I'm an idiot: I have a good degree in mathematics from the University of Oxford and I've been programming commercially for almost 12 years and in Scala for about a year (also commercially). I have just started to look at the Scala collections library re-implementation which is coming in the imminent 2.8 release. Those familiar with the library from 2.7 will notice that the library, from a usage perspective, has changed little. For example... > List("Paris", "London").map(_.length) res0: List[Int] List(5, 6) ...would work in either versions. The library is eminently useable: in fact it's fantastic. However, those previously unfamiliar with Scala and poking around to get a feel for the language now have to make sense of method signatures like: def map[B, That](f: A => B)(implicit bf: CanBuildFrom[Repr, B, That]): That For such simple functionality, this is a daunting signature and one which I find myself struggling to understand. Not that I think Scala was ever likely to be the next Java (or /C/C++/C#) - I don't believe its creators were aiming it at that market - but I think it is/was certainly feasible for Scala to become the next Ruby or Python (i.e. to gain a significant commercial user-base) Is this going to put people off coming to Scala? Is this going to give Scala a bad name in the commercial world as an academic plaything that only dedicated PhD students can understand? Are CTOs and heads of software going to get scared off? Was the library re-design a sensible idea? If you're using Scala commercially, are you worried about this? Are you planning to adopt 2.8 immediately or wait to see what happens? Steve Yegge once attacked Scala (mistakenly in my opinion) for what he saw as its overcomplicated type-system. I worry that someone is going to have a field day spreading fud with this API (similarly to how Josh Bloch scared the JCP out of adding closures to Java). Note - I should be clear that, whilst I believe that Josh Bloch was influential in the rejection of the BGGA closures proposal, I don't ascribe this to anything other than his honestly-held beliefs that the proposal represented a mistake.

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  • physics game programming box2d - orientating a turret-like object using torques

    - by egarcia
    This is a problem I hit when trying to implement a game using the LÖVE engine, which covers box2d with Lua scripting. The objective is simple: A turret-like object (seen from the top, on a 2D environment) needs to orientate itself so it points to a target. The turret is on the x,y coordinates, and the target is on tx, ty. We can consider that x,y are fixed, but tx, ty tend to vary from one instant to the other (i.e. they would be the mouse cursor). The turret has a rotor that can apply a rotational force (torque) on any given moment, clockwise or counter-clockwise. The magnitude of that force has an upper limit called maxTorque. The turret also has certain rotational inertia, which acts for angular movement the same way mass acts for linear movement. There's no friction of any kind, so the turret will keep spinning if it has an angular velocity. The turret has a small AI function that re-evaluates its orientation to verify that it points to the right direction, and activates the rotator. This happens every dt (~60 times per second). It looks like this right now: function Turret:update(dt) local x,y = self:getPositon() local tx,ty = self:getTarget() local maxTorque = self:getMaxTorque() -- max force of the turret rotor local inertia = self:getInertia() -- the rotational inertia local w = self:getAngularVelocity() -- current angular velocity of the turret local angle = self:getAngle() -- the angle the turret is facing currently -- the angle of the like that links the turret center with the target local targetAngle = math.atan2(oy-y,ox-x) local differenceAngle = _normalizeAngle(targetAngle - angle) if(differenceAngle <= math.pi) then -- counter-clockwise is the shortest path self:applyTorque(maxTorque) else -- clockwise is the shortest path self:applyTorque(-maxTorque) end end ... it fails. Let me explain with two illustrative situations: The turret "oscillates" around the targetAngle. If the target is "right behind the turret, just a little clock-wise", the turret will start applying clockwise torques, and keep applying them until the instant in which it surpasses the target angle. At that moment it will start applying torques on the opposite direction. But it will have gained a significant angular velocity, so it will keep going clockwise for some time... until the target will be "just behind, but a bit counter-clockwise". And it will start again. So the turret will oscillate or even go in round circles. I think that my turret should start applying torques in the "opposite direction of the shortest path" before it reaches the target angle (like a car braking before stopping). Intuitively, I think the turret should "start applying torques on the opposite direction of the shortest path when it is about half-way to the target objective". My intuition tells me that it has something to do with the angular velocity. And then there's the fact that the target is mobile - I don't know if I should take that into account somehow or just ignore it. How do I calculate when the turret must "start braking"?

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  • Should I use Python or Assembly for a super fast copy program

    - by PyNEwbie
    As a maintenance issue I need to routinely (3-5 times per year) copy a repository that is now has over 20 million files and exceeds 1.5 terabytes in total disk space. I am currently using RICHCOPY, but have tried others. RICHCOPY seems the fastest but I do not believe I am getting close to the limits of the capabilities of my XP machine. I am toying around with using what I have read in The Art of Assembly Language to write a program to copy my files. My other thought is to start learning how to multi-thread in Python to do the copies. I am toying around with the idea of doing this in Assembly because it seems interesting, but while my time is not incredibly precious it is precious enough that I am trying to get a sense of whether or not I will see significant enough gains in copy speed. I am assuming that I would but I only started really learning to program 18 months and it is still more or less a hobby. Thus I may be missing some fundamental concept of what happens with interpreted languages. Any observations or experiences would be appreciated. Note, I am not looking for any code. I have already written a basic copy program in Python 2.6 that is no slower than RICHCOPY. I am looking for some observations on which will give me more speed. Right now it takes me over 50 hours to make a copy from a disk to a Drobo and then back from the Drobo to a disk. I have a LogicCube for when I am simply duplicating a disk but sometimes I need to go from a disk to Drobo or the reverse. I am thinking that given that I can sector copy a 3/4 full 2 terabyte drive using the LogicCube in under seven hours I should be able to get close to that using Assembly, but I don't know enough to know if this is valid. (Yes, sometimes ignorance is bliss) The reason I need to speed it up is I have had two or three cycles where something has happened during copy (fifty hours is a long time to expect the world to hold still) that has caused me to have to trash the copy and start over. For example, last week the water main broke under our building and shorted out the power.

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  • "Emulating" Application.Run using Application.DoEvents

    - by Luca
    I'm getting in trouble. I'm trying to emulate the call Application.Run using Application.DoEvents... this sounds bad, and then I accept also alternative solutions to my question... I have to handle a message pump like Application.Run does, but I need to execute code before and after the message handling. Here is the main significant snippet of code. // Create barrier (multiple kernels synchronization) sKernelBarrier = new KernelBarrier(sKernels.Count); foreach (RenderKernel k in sKernels) { // Create rendering contexts (one for each kernel) k.CreateRenderContext(); // Start render kernel kernels k.mThread = new Thread(RenderKernelMain); k.mThread.Start(k); } while (sKernelBarrier.KernelCount > 0) { // Wait untill all kernel loops has finished sKernelBarrier.WaitKernelBarrier(); // Do application events Application.DoEvents(); // Execute shared context services foreach (RenderKernelContextService s in sContextServices) s.Execute(sSharedContext); // Next kernel render loop sKernelBarrier.ReleaseKernelBarrier(); } This snippet of code is execute by the Main routine. Pratically I have a list of Kernel classes, which runs in separate threads, these threads handle a Form for rendering in OpenGL. I need to synchronize all the Kernel threads using a barrier, and this work perfectly. Of course, I need to handle Form messages in the main thread (Main routine), for every Form created, and indeed I call Application.DoEvents() to do the job. Now I have to modify the snippet above to have a common Form (simple dialog box) without consuming the 100% of CPU calling Application.DoEvents(), as Application.Run does. The goal should be to have the snippet above handle messages when arrives, and issue a rendering (releasing the barrier) only when necessary, without trying to get the maximum FPS; there should be the possibility to switch to a strict loop to render as much as possible. How could it be possible? Note: the snippet above must be executed in the Main routine, since the OpenGL context is created on the main thread. Moving the snippet in a separated thread and calling Application.Run is quite unstable and buggy...

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  • How to make freelance clients understand the costs of developing and maintaining mature products?

    - by John
    I have a freelance web application project where the client requests new features every two weeks or so. I am unable to anticipate the requirements of upcoming features. So when the client requests a new feature, one of several things may happen: I implement the feature with ease because it is compatible with the existing platform I implement the feature with difficulty because I have to rewrite a significant portion of the platform's foundation Client withdraws request because it costs too much to implement against existing platform At the beginning of the project, for about six months, all feature requests fell under category 1) because the system was small and agile. But for the past six months, most feature implementation fell under category 2). The system is mature, forcing me to refactor and test everytime I want to add new modules. Additionally, I find myself breaking things that use to work, and fixing it (I don't get paid for this). The client is starting to express frustration at the time and cost for me to implement new features. To them, many of the feature requests are of the same scale as the features they requested six months ago. For example, a client would ask, "If it took you 1 week to build a ticketing system last year, why does it take you 1 month to build an event registration system today? An event registration system is much simpler than a ticketing system. It should only take you 1 week!" Because of this scenario, I fear feature requests will soon land in category 3). In fact, I'm already eating a lot of the cost myself because I volunteer many hours to support the project. The client is often shocked when I tell him honestly the time it takes to do something. The client always compares my estimates against the early months of a project. I don't think they're prepared for what it really costs to develop, maintain and support a mature web application. When working on a salary for a full time company, managers were more receptive of my estimates and even encouraged me to pad my numbers to prepare for the unexpected. Is there a way to condition my clients to think the same way? Can anyone offer advice on how I can continue to work on this web project without eating too much of the cost myself? Additional info - I've only been freelancing full time for 1 year. I don't yet have the high end clients, but I'm slowly getting there. I'm getting better quality clients as time goes by.

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  • Grouping geographical shapes

    - by grenade
    I am using Dundas Maps and attempting to draw a map of the world where countries are grouped into regions that are specific to a business implementation. I have shape data (points and segments) for each country in the world. I can combine countries into regions by adding all points and segments for countries within a region to a new region shape. foreach(var region in GetAllRegions()){ var regionShape = new Shape { Name = region.Name }; foreach(var country in GetCountriesInRegion(region.Id)){ var countryShape = GetCountryShape(country.Id); regionShape.AddSegments(countryShape.ShapeData.Points, countryShape.ShapeData.Segments); } map.Shapes.Add(regionShape); } The problem is that the country border lines still show up within a region and I want to remove them so that only regional borders show up. Dundas polygons must start and end at the same point. This is the case for all the country shapes. Now I need an algorithm that can: Determine where country borders intersect at a regional border, so that I can join the regional border segments. Determine which country borders are not regional borders so that I can discard them. Sort the resulting regional points so that they sequentialy describe the shape boundaries. Below is where I have gotten to so far with the map. You can see that the country borders still need to be removed. For example, the border between Mongolia and China should be discarded whereas the border between Mongolia and Russia should be retained. The reason I need to retain a regional border is that the region colors will be significant in conveying information but adjacent regions may be the same color. The regions can change to include or exclude countries and this is why the regional shaping must be dynamic. EDIT: I now know that I what I am looking for is a UNION of polygons. David Lean explains how to do it using the spatial functions in SQL Server 2008 which might be an option but my efforts have come to a halt because the resulting polygon union is so complex that SQL truncates it at 43,680 characters. I'm now trying to either find a workaround for that or find a way of doing the union in code.

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  • R glm standard error estimate differences to SAS PROC GENMOD

    - by Michelle
    I am converting a SAS PROC GENMOD example into R, using glm in R. The SAS code was: proc genmod data=data0 namelen=30; model boxcoxy=boxcoxxy ~ AGEGRP4 + AGEGRP5 + AGEGRP6 + AGEGRP7 + AGEGRP8 + RACE1 + RACE3 + WEEKEND + SEQ/dist=normal; FREQ REPLICATE_VAR; run; My R code is: parmsg2 <- glm(boxcoxxy ~ AGEGRP4 + AGEGRP5 + AGEGRP6 + AGEGRP7 + AGEGRP8 + RACE1 + RACE3 + WEEKEND + SEQ , data=data0, family=gaussian, weights = REPLICATE_VAR) When I use summary(parmsg2) I get the same coefficient estimates as in SAS, but my standard errors are wildly different. The summary output from SAS is: Name df Estimate StdErr LowerWaldCL UpperWaldCL ChiSq ProbChiSq Intercept 1 6.5007436 .00078884 6.4991975 6.5022897 67911982 0 agegrp4 1 .64607262 .00105425 .64400633 .64813891 375556.79 0 agegrp5 1 .4191395 .00089722 .41738099 .42089802 218233.76 0 agegrp6 1 -.22518765 .00083118 -.22681672 -.22355857 73401.113 0 agegrp7 1 -1.7445189 .00087569 -1.7462352 -1.7428026 3968762.2 0 agegrp8 1 -2.2908855 .00109766 -2.2930369 -2.2887342 4355849.4 0 race1 1 -.13454883 .00080672 -.13612997 -.13296769 27817.29 0 race3 1 -.20607036 .00070966 -.20746127 -.20467944 84319.131 0 weekend 1 .0327884 .00044731 .0319117 .03366511 5373.1931 0 seq2 1 -.47509583 .00047337 -.47602363 -.47416804 1007291.3 0 Scale 1 2.9328613 .00015586 2.9325559 2.9331668 -127 The summary output from R is: Coefficients: Estimate Std. Error t value Pr(>|t|) (Intercept) 6.50074 0.10354 62.785 < 2e-16 AGEGRP4 0.64607 0.13838 4.669 3.07e-06 AGEGRP5 0.41914 0.11776 3.559 0.000374 AGEGRP6 -0.22519 0.10910 -2.064 0.039031 AGEGRP7 -1.74452 0.11494 -15.178 < 2e-16 AGEGRP8 -2.29089 0.14407 -15.901 < 2e-16 RACE1 -0.13455 0.10589 -1.271 0.203865 RACE3 -0.20607 0.09315 -2.212 0.026967 WEEKEND 0.03279 0.05871 0.558 0.576535 SEQ -0.47510 0.06213 -7.646 2.25e-14 The importance of the difference in the standard errors is that the SAS coefficients are all statistically significant, but the RACE1 and WEEKEND coefficients in the R output are not. I have found a formula to calculate the Wald confidence intervals in R, but this is pointless given the difference in the standard errors, as I will not get the same results. Apparently SAS uses a ridge-stabilized Newton-Raphson algorithm for its estimates, which are ML. The information I read about the glm function in R is that the results should be equivalent to ML. What can I do to change my estimation procedure in R so that I get the equivalent coefficents and standard error estimates that were produced in SAS? To update, thanks to Spacedman's answer, I used weights because the data are from individuals in a dietary survey, and REPLICATE_VAR is a balanced repeated replication weight, that is an integer (and quite large, in the order of 1000s or 10000s). The website that describes the weight is here. I don't know why the FREQ rather than the WEIGHT command was used in SAS. I will now test by expanding the number of observations using REPLICATE_VAR and rerunning the analysis.

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  • Java map / nio / NFS issue causing a VM fault: "a fault occurred in a recent unsafe memory access op

    - by Matthew Bloch
    I have written a parser class for a particular binary format (nfdump if anyone is interested) which uses java.nio's MappedByteBuffer to read through files of a few GB each. The binary format is just a series of headers and mostly fixed-size binary records, which are fed out to the called by calling nextRecord(), which pushes on the state machine, returning null when it's done. It performs well. It works on a development machine. On my production host, it can run for a few minutes or hours, but always seems to throw "java.lang.InternalError: a fault occurred in a recent unsafe memory access operation in compiled Java code", fingering one of the Map.getInt, getShort methods, i.e. a read operation in the map. The uncontroversial (?) code that sets up the map is this: /** Set up the map from the given filename and position */ protected void open() throws IOException { // Set up buffer, is this all the flexibility we'll need? channel = new FileInputStream(file).getChannel(); MappedByteBuffer map1 = channel.map(FileChannel.MapMode.READ_ONLY, 0, channel.size()); map1.load(); // we want the whole thing, plus seems to reduce frequency of crashes? map = map1; // assumes the host writing the files is little-endian (x86), ought to be configurable map.order(java.nio.ByteOrder.LITTLE_ENDIAN); map.position(position); } and then I use the various map.get* methods to read shorts, ints, longs and other sequences of bytes, before hitting the end of the file and closing the map. I've never seen the exception thrown on my development host. But the significant point of difference between my production host and development is that on the former, I am reading sequences of these files over NFS (probably 6-8TB eventually, still growing). On my dev machine, I have a smaller selection of these files locally (60GB), but when it blows up on the production host it's usually well before it gets to 60GB of data. Both machines are running java 1.6.0_20-b02, though the production host is running Debian/lenny, the dev host is Ubuntu/karmic. I'm not convinced that will make any difference. Both machines have 16GB RAM, and are running with the same java heap settings. I take the view that if there is a bug in my code, there is enough of a bug in the JVM not to throw me a proper exception! But I think it is just a particular JVM implementation bug due to interactions between NFS and mmap, possibly a recurrence of 6244515 which is officially fixed. I already tried adding in a "load" call to force the MappedByteBuffer to load its contents into RAM - this seemed to delay the error in the one test run I've done, but not prevent it. Or it could be coincidence that was the longest it had gone before crashing! If you've read this far and have done this kind of thing with java.nio before, what would your instinct be? Right now mine is to rewrite it without nio :)

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  • .Net Dynamically Load DLL

    - by hermiod
    I am trying to write some code that will allow me to dynamically load DLLs into my application, depending on an application setting. The idea is that the database to be accessed is set in the application settings and then this loads the appropriate DLL and assigns it to an instance of an interface for my application to access. This is my code at the moment: Dim SQLDataSource As ICRDataLayer Dim ass As Assembly = Assembly. _ LoadFrom("M:\MyProgs\WebService\DynamicAssemblyLoading\SQLServer\bin\Debug\SQLServer.dll") Dim obj As Object = ass.CreateInstance(GetType(ICRDataLayer).ToString, True) SQLDataSource = DirectCast(obj, ICRDataLayer) MsgBox(SQLDataSource.ModuleName & vbNewLine & SQLDataSource.ModuleDescription) I have my interface (ICRDataLayer) and the SQLServer.dll contains an implementation of this interface. I just want to load the assembly and assign it to the SQLDataSource object. The above code just doesn't work. There are no exceptions thrown, even the Msgbox doesn't appear. I would've expected at least the messagebox appearing with nothing in it, but even this doesn't happen! Is there a way to determine if the loaded assembly implements a specific interface. I tried the below but this also doesn't seem to do anything! For Each loadedType As Type In ass.GetTypes If GetType(ICRDataLayer).IsAssignableFrom(loadedType) Then Dim obj1 As Object = ass.CreateInstance(GetType(ICRDataLayer).ToString, True) SQLDataSource = DirectCast(obj1, ICRDataLayer) End If Next EDIT: New code from Vlad's examples: Module CRDataLayerFactory Sub New() End Sub ' class name is a contract, ' should be the same for all plugins Private Function Create() As ICRDataLayer Return New SQLServer() End Function End Module Above is Module in each DLL, converted from Vlad's C# example. Below is my code to bring in the DLL: Dim SQLDataSource As ICRDataLayer Dim ass As Assembly = Assembly. _ LoadFrom("M:\MyProgs\WebService\DynamicAssemblyLoading\SQLServer\bin\Debug\SQLServer.dll") Dim factory As Object = ass.CreateInstance("CRDataLayerFactory", True) Dim t As Type = factory.GetType Dim method As MethodInfo = t.GetMethod("Create") Dim obj As Object = method.Invoke(factory, Nothing) SQLDataSource = DirectCast(obj, ICRDataLayer) EDIT: Implementation based on Paul Kohler's code Dim file As String For Each file In Directory.GetFiles(baseDir, searchPattern, SearchOption.TopDirectoryOnly) Dim assemblyType As System.Type For Each assemblyType In Assembly.LoadFrom(file).GetTypes Dim s As System.Type() = assemblyType.GetInterfaces For Each ty As System.Type In s If ty.Name.Contains("ICRDataLayer") Then MsgBox(ty.Name) plugin = DirectCast(Activator.CreateInstance(assemblyType), ICRDataLayer) MessageBox.Show(plugin.ModuleName) End If Next I get the following error with this code: Unable to cast object of type 'SQLServer.CRDataSource.SQLServer' to type 'DynamicAssemblyLoading.ICRDataLayer'. The actual DLL is in a different project called SQLServer in the same solution as my implementation code. CRDataSource is a namespace and SQLServer is the actual class name of the DLL. The SQLServer class implements ICRDataLayer, so I don't understand why it wouldn't be able to cast it. Is the naming significant here, I wouldn't have thought it would be.

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  • How to force VS 2010 to skip "builds" of projects which haven't changed?

    - by Ladislav Mrnka
    Our product's solution has more than 100+ projects (500+ksloc of production code). Most of them are C# projects but we also have few using C++/CLI to bridge communication with native code. Rebuilding the whole solution takes several minutes. That's fine. If I want to rebuilt the solution I expect that it will really take some time. What is not fine is time needed to build solution after full rebuild. Imagine I used full rebuild and know without doing any changes to to the solution I press Build (F6 or Ctrl+Shift+B). Why it takes 35s if there was no change? In output I see that it started "building" of each project - it doesn't perform real build but it does something which consumes significant amount of time. That 35s delay is pain in the ass. Yes I can improve the time by not using build solution but only build project (Shift+F6). If I run build project on particular test project I'm currently working on it will take "only" 8+s. It requires me to run project build on correct project (the test project to ensure dependent tested code is build as well). At least ReSharper test runner correctly recognizes that only this single project must be build and rerunning test usually contains only 8+s compilation. My current coding Kata is: don't touch Ctrl+Shift+B. The test project build will take 8s even if I don't do any changes. The reason why it takes 8s is because it also "builds" dependencies = in my case it "builds" more than 20 projects but I made changes only to unit test or single dependency! I don't want it to touch other projects. Is there a way to simply tell VS to build only projects where some changes were done and projects which are dependent on changed ones (preferably this part as another build option)? I worry you will tell me that it is exactly what VS is doing but in MS way ... I want to improve my TDD experience and reduce the time of compilation (in TDD the compilation can happen twice per minute). To make this even more frustrated I'm working in a team where most of developers used to work on Java projects prior to joining this one. So you can imagine how they are pissed off when they must use VS in contrast to full incremental compilation in Java. I don't require incremental compilation of classes. I expect working incremental compilation of solutions. Especially in product like VS 2010 Ultimate which costs several thousands dollars. I really don't want to get answers like: Make a separate solution Unload projects you don't need etc. I can read those answers here. Those are not acceptable solutions. We're not paying for VS to do such compromises.

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  • CUDA, more threads for same work = Longer run time despite better occupancy, Why?

    - by zenna
    I encountered a strange problem where increasing my occupancy by increasing the number of threads reduced performance. I created the following program to illustrate the problem: #include <stdio.h> #include <stdlib.h> #include <cuda_runtime.h> __global__ void less_threads(float * d_out) { int num_inliers; for (int j=0;j<800;++j) { //Do 12 computations num_inliers += threadIdx.x*1; num_inliers += threadIdx.x*2; num_inliers += threadIdx.x*3; num_inliers += threadIdx.x*4; num_inliers += threadIdx.x*5; num_inliers += threadIdx.x*6; num_inliers += threadIdx.x*7; num_inliers += threadIdx.x*8; num_inliers += threadIdx.x*9; num_inliers += threadIdx.x*10; num_inliers += threadIdx.x*11; num_inliers += threadIdx.x*12; } if (threadIdx.x == -1) d_out[blockIdx.x*blockDim.x+threadIdx.x] = num_inliers; } __global__ void more_threads(float *d_out) { int num_inliers; for (int j=0;j<800;++j) { // Do 4 computations num_inliers += threadIdx.x*1; num_inliers += threadIdx.x*2; num_inliers += threadIdx.x*3; num_inliers += threadIdx.x*4; } if (threadIdx.x == -1) d_out[blockIdx.x*blockDim.x+threadIdx.x] = num_inliers; } int main(int argc, char* argv[]) { float *d_out = NULL; cudaMalloc((void**)&d_out,sizeof(float)*25000); more_threads<<<780,128>>>(d_out); less_threads<<<780,32>>>(d_out); return 0; } Note both kernels should do the same amount of work in total, the (if threadIdx.x == -1 is a trick to stop the compiler optimising everything out and leaving an empty kernel). The work should be the same as more_threads is using 4 times as many threads but with each thread doing 4 times less work. Significant results form the profiler results are as followsL: more_threads: GPU runtime = 1474 us,reg per thread = 6,occupancy=1,branch=83746,divergent_branch = 26,instructions = 584065,gst request=1084552 less_threads: GPU runtime = 921 us,reg per thread = 14,occupancy=0.25,branch=20956,divergent_branch = 26,instructions = 312663,gst request=677381 As I said previously, the run time of the kernel using more threads is longer, this could be due to the increased number of instructions. Why are there more instructions? Why is there any branching, let alone divergent branching, considering there is no conditional code? Why are there any gst requests when there is no global memory access? What is going on here! Thanks

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  • should I ever put a major version number into a C#/Java namespace?

    - by Andrew Patterson
    I am designing a set of 'service' layer objects (data objects and interface definitions) for a WCF web service (that will be consumed by third party clients i.e. not in-house, so outside my direct control). I know that I am not going to get the interface definition exactly right - and am wanting to prepare for the time when I know that I will have to introduce a breaking set of new data objects. However, the reality of the world I am in is that I will also need to run my first version simultaneously for quite a while. The first version of my service will have URL of http://host/app/v1service.svc and when the times comes by new version will live at http://host/app/v2service.svc However, when it comes to the data objects and interfaces, I am toying with putting the 'major' version of the interface number into the actual namespace of the classes. namespace Company.Product.V1 { [DataContract(Namespace = "company-product-v1")] public class Widget { [DataMember] string widgetName; } public interface IFunction { Widget GetWidgetData(int code); } } When the time comes for a fundamental change to the service, I will introduce some classes like namespace Company.Product.V2 { [DataContract(Namespace = "company-product-v2")] public class Widget { [DataMember] int widgetCode; [DataMember] int widgetExpiry; } public interface IFunction { Widget GetWidgetData(int code); } } The advantages as I see it are that I will be able to have a single set of code serving both interface versions, sharing functionality where possible. This is because I will be able to reference both interface versions as a distinct set of C# objects. Similarly, clients may use both interface versions simultaneously, perhaps using V1.Widget in some legacy code whilst new bits move on to V2.Widget. Can anyone tell why this is a stupid idea? I have a nagging feeling that this is a bit smelly.. notes: I am obviously not proposing every single new version of the service would be in a new namespace. Presumably I will do as many non-breaking interface changes as possible, but I know that I will hit a point where all the data modelling will probably need a significant rewrite. I understand assembly versioning etc but I think this question is tangential to that type of versioning. But I could be wrong.

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  • Is there a scheduling algorithm that optimizes for "maker's schedules"?

    - by John Feminella
    You may be familiar with Paul Graham's essay, "Maker's Schedule, Manager's Schedule". The crux of the essay is that for creative and technical professionals, meetings are anathema to productivity, because they tend to lead to "schedule fragmentation", breaking up free time into chunks that are too small to acquire the focus needed to solve difficult problems. In my firm we've seen significant benefits by minimizing the amount of disruption caused, but the brute-force algorithm we use to decide schedules is not sophisticated enough to handle scheduling large groups of people well. (*) What I'm looking for is if there's are any well-known algorithms which minimize this productivity disruption, among a group of N makers and managers. In our model, There are N people. Each person pi is either a maker (Mk) or a manager (Mg). Each person has a schedule si. Everyone's schedule is H hours long. A schedule consists of a series of non-overlapping intervals si = [h1, ..., hj]. An interval is either free or busy. Two adjacent free intervals are equivalent to a single free interval that spans both. A maker's productivity is maximized when the number of free intervals is minimized. A manager's productivity is maximized when the total length of free intervals is maximized. Notice that if there are no meetings, both the makers and the managers experience optimum productivity. If meetings must be scheduled, then makers prefer that meetings happen back-to-back, while managers don't care where the meeting goes. Note that because all disruptions are treated as equally harmful to makers, there's no difference between a meeting that lasts 1 second and a meeting that lasts 3 hours if it segments the available free time. The problem is to decide how to schedule M different meetings involving arbitrary numbers of the N people, where each person in a given meeting must place a busy interval into their schedule such that it doesn't overlap with any other busy interval. For each meeting Mt the start time for the busy interval must be the same for all parties. Does an algorithm exist to solve this problem or one similar to it? My first thought was that this looks really similar to defragmentation (minimize number of distinct chunks), and there are a lot of algorithms about that. But defragmentation doesn't have much to do with scheduling. Thoughts? (*) Practically speaking this is not really a problem, because it's rare that we have meetings with more than ~5 people at once, so the space of possibilities is small.

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  • Finding N contiguous zero bits in an integer to the left of the MSB position of another integer

    - by James Morris
    The problem is: given an integer val1 find the position of the highest bit set (Most Significant Bit) then, given a second integer val2 find a contiguous region of unset bits, with the minimum number of zero bits given by width to the left of the position (ie, in the higher bits). Here is the C code for my solution: typedef unsigned int t; unsigned const t_bits = sizeof(t) * CHAR_BIT; _Bool test_fit_within_left_of_msb( unsigned width, t val1, t val2, unsigned* offset_result) { unsigned offbit = 0; unsigned msb = 0; t mask; t b; while(val1 >>= 1) ++msb; while(offbit + width < t_bits - msb) { mask = (((t)1 << width) - 1) << (t_bits - width - offbit); b = val2 & mask; if (!b) { *offset_result = offbit; return true; } if (offbit++) /* this conditional bothers me! */ b <<= offbit - 1; while(b <<= 1) offbit++; } return false; } Aside from faster ways of finding the MSB of the first integer, the commented test for a zero offbit seems a bit extraneous, but necessary to skip the highest bit of type t if it is set. I have also implemented similar algorithms but working to the right of the MSB of the first number, so they don't require this seemingly extra condition. How can I get rid of this extra condition, or even, are there far more optimal solutions? Edit: Some background not strictly required. The offset result is a count of bits from the high bit, not from the low bit as maybe expected. This will be part of a wider algorithm which scans a 2D array for a 2D area of zero bits. Here, for testing, the algorithm has been simplified. val1 represents the first integer which does not have all bits set found in a row of the 2D array. From this the 2D version would scan down which is what val2 represents. Here's some output showing success and failure: t_bits:32 t_high: 10000000000000000000000000000000 ( 2147483648 ) --------- ----------------------------------- *** fit within left of msb test *** ----------------------------------- val1: 00000000000000000000000010000000 ( 128 ) val2: 01000001000100000000100100001001 ( 1091569929 ) msb: 7 offbit:0 + width: 8 = 8 mask: 11111111000000000000000000000000 ( 4278190080 ) b: 01000001000000000000000000000000 ( 1090519040 ) offbit:8 + width: 8 = 16 mask: 00000000111111110000000000000000 ( 16711680 ) b: 00000000000100000000000000000000 ( 1048576 ) offbit:12 + width: 8 = 20 mask: 00000000000011111111000000000000 ( 1044480 ) b: 00000000000000000000000000000000 ( 0 ) offbit:12 iters:10 ***** found room for width:8 at offset: 12 ***** ----------------------------------- *** fit within left of msb test *** ----------------------------------- val1: 00000000000000000000000001000000 ( 64 ) val2: 00010000000000001000010001000001 ( 268469313 ) msb: 6 offbit:0 + width: 13 = 13 mask: 11111111111110000000000000000000 ( 4294443008 ) b: 00010000000000000000000000000000 ( 268435456 ) offbit:4 + width: 13 = 17 mask: 00001111111111111000000000000000 ( 268402688 ) b: 00000000000000001000000000000000 ( 32768 ) ***** mask: 00001111111111111000000000000000 ( 268402688 ) offbit:17 iters:15 ***** no room found for width:13 ***** (iters is the count of iterations of the inner while loop)

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  • Performing full screen grab in windows

    - by Steven Lu
    I am working an idea that involves getting a full capture of the screen including windows and apps, analyzing it, and then drawing items back onto the screen, as an overlay. I want to learn image processing techniques and I could get lots of data to work with if I can directly access the Windows screen. I could use this to build automation tools the likes of which have never been seen before. More on that later. I have full screen capture working for the most part. HWND hwind = GetDesktopWindow(); HDC hdc = GetDC(hwind); int resx = GetSystemMetrics(SM_CXSCREEN); int resy = GetSystemMetrics(SM_CYSCREEN); int BitsPerPixel = GetDeviceCaps(hdc,BITSPIXEL); HDC hdc2 = CreateCompatibleDC(hdc); BITMAPINFO info; info.bmiHeader.biSize = sizeof(BITMAPINFOHEADER); info.bmiHeader.biWidth = resx; info.bmiHeader.biHeight = resy; info.bmiHeader.biPlanes = 1; info.bmiHeader.biBitCount = BitsPerPixel; info.bmiHeader.biCompression = BI_RGB; void *data; hbitmap = CreateDIBSection(hdc2,&info,DIB_RGB_COLORS,(void**)&data,0,0); SelectObject(hdc2,hbitmap); Once this is done, I can call this repeatedly: BitBlt(hdc2,0,0,resx,resy,hdc,0,0,SRCCOPY); The cleanup code (I have no idea if this is correct): DeleteObject(hbitmap); ReleaseDC(hwind,hdc); if (hdc2) { DeleteDC(hdc2); } Every time BitBlt is called it grabs the screen and saves it in memory I can access thru data. Performance is somewhat satisfactory. BitBlt executes in 50 milliseconds (sometimes as low as 33ms) at 1920x1200x32. What surprises me is that when I switch display mode to 16 bit, 1920x1200x16, either through my graphics settings beforehand, or by using ChangeDisplaySettings, I get a massively improved screen grab time between 1ms and 2ms, which cannot be explained by the factor of two reduction in bit-depth. Using CreateDIBSection (as above) offers a significant speed up when in 16-bit mode, compared to if I set up with CreateCompatibleBitmap (6-7ms/f). Does anybody know why dropping to 16bit causes such a speed increase? Is there any hope for me to grab 32bit at such speeds? if not for the color depth, but for not forcing a change of screen buffer modes and the awful flickering.

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  • how to develop a program to minimize errors in human transcription of hand written surveys

    - by Alex. S.
    I need to develop custom software to do surveys. Questions may be of multiple choice, or free text in a very few cases. I was asked to design a subsystem to check if there is any error in the manual data entry for the multiple choices part. We're trying to speed up the user data entry process and to minimize human input differences between digital forms and the original questionnaires. The surveys are filled with handwritten marks and text by human interviewers, so it's possible to find hard to read marks, or also the user could accidentally select a different value in some question, and we would like to avoid that. The software must include some automatic control to detect possible typing differences. Each answer of the multiple choice questions has the same probability of being selected. This question has two parts: The GUI. The most simple thing I have in mind is to implement the most usable design of the questions display: use of large and readable fonts and space generously the choices. Is there something else? For faster input, I would like to use drop down lists (favoring keyboard over mouse). Given the questions are grouped in sections, I would like to show the answers selected for the questions of that section, but this could slow down the process. Any other ideas? The error checking subsystem. What else can I do to minimize or to check human typos in the multiple choice questions? Is this a solvable problem? is there some statistical methodology to check values that were entered by the users are the same from the hand filled forms? For example, let's suppose the survey has 5 questions, and each has 4 options. Let's say I have n survey forms filled in paper by interviewers, and they're ready to be entered in the software, then how to minimize the accidental differences that can have the manual transcription of the n surveys, without having to double check everything in the 5 questions of the n surveys? My first suggestion is that at the end of the processing of all the hand filled forms, the software could choose some forms randomly to make a double check of the responses in a few instances, but on what criteria can I make this selection? This validation would be enough to cover everything in a significant way? The actual survey is nation level and it has 56 pages with over 200 questions in total, so it will be a lot of hand written pages by many people, and the intention is to reduce the likelihood of errors and to optimize speed in the data entry process. The surveys must filled in paper first, given the complications of taking laptops or handhelds with the interviewers.

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  • Supporting multiple instances of a plugin DLL with global data

    - by Bruno De Fraine
    Context: I converted a legacy standalone engine into a plugin component for a composition tool. Technically, this means that I compiled the engine code base to a C DLL which I invoke from a .NET wrapper using P/Invoke; the wrapper implements an interface defined by the composition tool. This works quite well, but now I receive the request to load multiple instances of the engine, for different projects. Since the engine keeps the project data in a set of global variables, and since the DLL with the engine code base is loaded only once, loading multiple projects means that the project data is overwritten. I can see a number of solutions, but they all have some disadvantages: You can create multiple DLLs with the same code, which are seen as different DLLs by Windows, so their code is not shared. Probably this already works if you have multiple copies of the engine DLL with different names. However, the engine is invoked from the wrapper using DllImport attributes and I think the name of the engine DLL needs to be known when compiling the wrapper. Obviously, if I have to compile different versions of the wrapper for each project, this is quite cumbersome. The engine could run as a separate process. This means that the wrapper would launch a separate process for the engine when it loads a project, and it would use some form of IPC to communicate with this process. While this is a relatively clean solution, it requires some effort to get working, I don't now which IPC technology would be best to set-up this kind of construction. There may also be a significant overhead of the communication: the engine needs to frequently exchange arrays of floating-point numbers. The engine could be adapted to support multiple projects. This means that the global variables should be put into a project structure, and every reference to the globals should be converted to a corresponding reference that is relative to a particular project. There are about 20-30 global variables, but as you can imagine, these global variables are referenced from all over the code base, so this conversion would need to be done in some automatic manner. A related problem is that you should be able to reference the "current" project structure in all places, but passing this along as an extra argument in each and every function signature is also cumbersome. Does there exist a technique (in C) to consider the current call stack and find the nearest enclosing instance of a relevant data value there? Can the stackoverflow community give some advice on these (or other) solutions?

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  • Is there a way to keep track of the ordering of items in a dictionary?

    - by Corpsekicker
    I have a Dictionary<Guid, ElementViewModel>. (ElementViewModel is our own complex type.) I add items to the dictionary with a stock standard items.Add(Guid.NewGuid, new ElementViewModel() { /*setters go here*/ });, At a later stage I remove some or all of these items. A simplistic view of my ElementViewModel is this: class ElementViewModel { Guid Id { get; set; } string Name { get; set; } int SequenceNo { get; set; } } It may be significant to mention that the SequenceNos are compacted within the collection after adding, in case other operations like moving and copying took place. {1, 5, 6} - {1, 2, 3} A simplistic view of my remove operation is: public void RemoveElementViewModel(IEnumerable<ElementViewModel> elementsToDelete) { foreach (var elementViewModel in elementsToDelete) items.Remove(elementViewModel.Id); CompactSequenceNumbers(); } I will illustrate the problem with an example: I add 3 items to the dictionary: var newGuid = Guid.NewGuid(); items.Add(newGuid, new MineLayoutElementViewModel { Id = newGuid, SequenceNo = 1, Name = "Element 1" }); newGuid = Guid.NewGuid(); items.Add(newGuid, new MineLayoutElementViewModel { Id = newGuid, SequenceNo = 2, Name = "Element 2" }); newGuid = Guid.NewGuid(); items.Add(newGuid, new MineLayoutElementViewModel { Id = newGuid, SequenceNo = 3, Name = "Element 3" }); I remove 2 items RemoveElementViewModel(new List<ElementViewModel> { item2, item3 }); //imagine I had them cached somewhere. Now I want to add 2 other items: newGuid = Guid.NewGuid(); items.Add(newGuid, new MineLayoutElementViewModel { Id = newGuid, SequenceNo = 2, Name = "Element 2, Part 2" }); newGuid = Guid.NewGuid(); items.Add(newGuid, new MineLayoutElementViewModel { Id = newGuid, SequenceNo = 3, Name = "Element 3, Part 2" }); On evaluation of the dictionary at this point, I expected the order of items to be "Element 1", "Element 2, Part 2", "Element 3, Part 2" but it is actually in the following order: "Element 1", "Element 3, Part 2", "Element 2, Part 2" I rely on the order of these items to be a certain way. Why is it not as expected and what can I do about it?

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  • jQuery action being called when selector isn't met?

    - by dougoftheabaci
    I've been working on a prototype for a client's web site and I've run into a rather significant snag. You can view the prototype here. As you can see, the way it works is you can scroll a set of slides horizontally and, by clicking one, open a stack containing yet more slides. If you then click again on an image in that stack it opens up a lightbox. Clicking on another stack or the close button will close that stack (and open another, as case may be). That all works great. However you get some weird behavior if you do the following: Click to open any stack. Click to open an image's light box (this works best if you click on the image that's level with the main list). Close the light box and the stack either by clicking the close button or clicking on another stack. Click back to the first stack. Instead of reopening the stack, you get the lightbox. This confuses me as the light box should only ever be called if there is a class on the containing UL and that class is removed when the lightbox is closed. I've checked and double-checked this, it's definitely missing. Here are the respective functions: $("ul.hide a.lightbox").live("click",function(){ $("ul.show").removeClass("show").addClass("hide"); $(this).parent().parent().removeClass("hide").addClass("show"); $("ul.hide").animate({opacity: 0.2}); $("ul.show").animate({opacity: 1}); $("#next").animate({opacity: 0.2}); $("#prev").animate({opacity: 0.2}); return false; }); $("ul.show a.lightbox").live("click",function(){ $(this).fancybox().trigger("click"); return false; }); As you can see, in order for the lightbox to be called the containing UL has to have the class of show. However, if you check it with Firebug it won't. For those who are curious, the added .trigger("click"); is because the lightbox will require a double-click to launch otherwise. Any idea how I can fix this?

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  • php zencart mod - having problems with attributes array

    - by user80151
    I inherited a zencart mod and can't figure out what's wrong. The customer selects a product and an attribute (model#). This is then sent to another form that they complete. When they submit the form, the product and the attribute should be included in the email sent. At this time, only the product is coming through. The attribute just says "array." The interesting part is, when I delete the line that prints the attribute, the products_options_names will print out. So I know that both the product and the products_options_names are working. The attribute is the only thing that is not working right. Here's what I believe to be the significant code. This is the page that has the form, so the attribute should already be passed to the form. //Begin Adding of New features //$productsimage = $product['productsImage']; $productsname = $product['productsName']; $attributes = $product['attributes']; $products_options_name = $value['products_options_name']; $arr_product_list[] = "<strong>Product Name:</strong> $productsname <br />"; $arr_product_list[] .= "<strong>Attributes:</strong> $attributes <br />"; $arr_product_list[] .= "<strong>Products Options Name:</strong> $products_options_name <br />"; $arr_product_list[] .= "---------------------------------------------------------------"; //End Adding of New features } // end foreach ($productArray as $product) ?> Above this, there is another section that has attributes: <?php echo $product['attributeHiddenField']; if (isset($product['attributes']) && is_array($product['attributes'])) { echo '<div class="cartAttribsList">'; echo '<ul>'; reset($product['attributes']); foreach ($product['attributes'] as $option => $value) { ?> Can anyone help me figure out what is wrong? I'm not sure if the problem is on this page or if the attribute isn't being passed to this page. TIA

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  • Are there compelling reasons not to use Groovy?

    - by Leonard H Martin
    I'm developing a LoB application in Java after a long absence from the platform (having spent the last 8 years or so entrenched in Fortran, C, a smidgin of C++ and latterly .Net). Java, the language, is not much changed from how I remember it. I like it's strengths and I can work around its weaknesses - the platform has grown and deciding upon the myriad of different frameworks which appear to do much the same thing as one another is a different story; but that can wait for another day - all-in-all I'm comfortable with Java. However, over the last couple of weeks I've become enamoured with Groovy, and purely from a selfish point of view: but not just because it makes development against the JVM a more succinct and entertaining (and, well, "groovy") proposition than Java (the language). What strikes me most about Groovy is its inherent maintainability. We all (I hope!) strive to write well documented, easy to understand code. However, sometimes the languages we use themselves defeat us. An example: in 2001 I wrote a library in C to translate EDIFACT EDI messages into ANSI X12 messages. This is not a particularly complicated process, if slightly involved, and I thought at the time I had documented the code properly - and I probably had - but some six years later when I revisited the project (and after becoming acclimatised to C#) I found myself lost in so much C boilerplate (mallocs, pointers, etc. etc.) that it took three days of thoughtful analysis before I finally understood what I'd been doing six years previously. This evening I've written about 2000 lines of Java (it is the day of rest, after all!). I've documented as best as I know how, but, but, of those 2000 lines of Java a significant proportion is Java boiler plate. This is where I see Groovy and other dynamic languages winning through - maintainability and later comprehension. Groovy lets you concentrate on your intent without getting bogged down on the platform specific implementation; it's almost, but not quite, self documenting. I see this as being a huge boon to me when I revisit my current project (which I'll port to Groovy asap) in several years time and to my successors who will inherit it and carry on the good work. So, are there any reasons not to use Groovy?

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  • What IPC method should I use between Firefox extension and C# code running on the same machine?

    - by Rory
    I have a question about how to structure communication between a (new) Firefox extension and existing C# code. The firefox extension will use configuration data and will produce other data, so needs to get the config data from somewhere and save it's output somewhere. The data is produced/consumed by existing C# code, so I need to decide how the extension should interact with the C# code. Some pertinent factors: It's only running on windows, in a relatively controlled corporate environment. I have a windows service running on the machine, built in C#. Storing the data in a local datastore (like sqlite) would be useful for other reasons. The volume of data is low, e.g. 10kb of uncompressed xml every few minutes, and isn't very 'chatty'. The data exchange can be asynchronous for the most part if not completely. As with all projects, I have limited resources so want an option that's relatively easy. It doesn't have to be ultra-high performance, but shouldn't add significant overhead. I'm planning on building the extension in javascript (although could be convinced otherwise if really necessary) Some options I'm considering: use an XPCOM to .NET/COM bridge use a sqlite db: the extension would read from and save to it. The c# code would run in the service, populating the db and then processing data created by the service. use TCP sockets to communicate between the extension and the service. Let the service manage a local data store. My problem with (1) is I think this will be tricky and not so easy. But I could be completely wrong? The main problem I see with (2) is the locking of sqlite: only a single process can write data at a time so there'd be some blocking. However, it would be nice generally to have a local datastore so this is an attractive option if the performance impact isn't too great. I don't know whether (3) would be particularly easy or hard ... or what approach to take on the protocol: something custom or http. Any comments on these ideas or other suggestions? UPDATE: I was planning on building the extension in javascript rather than c++

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  • How can I improve my select query for storing large versioned data sets?

    - by Jason Francis
    At work, we build large multi-page web applications, consisting mostly of radio and check boxes. The primary purpose of each application is to gather data, but as users return to a page they have previously visited, we report back to them their previous responses. Worst-case scenario, we might have up to 900 distinct variables and around 1.5 million users. For several reasons, it makes sense to use an insert-only approach to storing the data (as opposed to update-in-place) so that we can capture historical data about repeated interactions with variables. The net result is that we might have several responses per user per variable. Our table to collect the responses looks something like this: CREATE TABLE [dbo].[results]( [id] [bigint] IDENTITY(1,1) NOT NULL, [userid] [int] NULL, [variable] [varchar](8) NULL, [value] [tinyint] NULL, [submitted] [smalldatetime] NULL) Where id serves as the primary key. Virtually every request results in a series of insert statements (one per variable submitted), and then we run a select to produce previous responses for the next page (something like this): SELECT t.id, t.variable, t.value FROM results t WITH (NOLOCK) WHERE t.userid = '2111846' AND (t.variable='internat' OR t.variable='veteran' OR t.variable='athlete') AND t.id IN (SELECT MAX(id) AS id FROM results WITH (NOLOCK) WHERE userid = '2111846' AND (t.variable='internat' OR t.variable='veteran' OR t.variable='athlete') GROUP BY variable) Which, in this case, would return the most recent responses for the variables "internat", "veteran", and "athlete" for user 2111846. We have followed the advice of the database tuning tools in indexing the tables, and against our data, this is the best-performing version of the select query that we have been able to come up with. Even so, there seems to be significant performance degradation as the table approaches 1 million records (and we might have about 150x that). We have a fairly-elegant solution in place for sharding the data across multiple tables which has been working quite well, but I am open for any advice about how I might construct a better version of the select query. We use this structure frequently for storing lots of independent data points, and we like the benefits it provides. So the question is, how can I improve the performance of the select query? I assume the nested select statement is a bad idea, but I have yet to find an alternative that performs as well. Thanks in advance. NB: Since we emphasize creating over reading in this case, and since we never update in place, there doesn't seem to be any penalty (and some advantage) for using the NOLOCK directive in this case.

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  • Determining if Memory Pointer is Valid - C++

    - by Jim Fell
    It has been my observation that if free( ptr ) is called where ptr is not a valid pointer to system-allocated memory, an access violation occurs. Let's say that I call free like this: LPVOID ptr = (LPVOID)0x12345678; free( ptr ); This will most definitely cause an access violation. Is there a way to test that the memory location pointed to by ptr is valid system-allocated memory? It seems to me that the the memory management part of the Windows OS kernel must know what memory has been allocated and what memory remains for allocation. Otherwise, how could it know if enough memory remains to satisfy a given request? (rhetorical) That said, it seems reasonable to conclude that there must be a function (or set of functions) that would allow a user to determine if a pointer is valid system-allocated memory. Perhaps Microsoft has not made these functions public. If Microsoft has not provided such an API, I can only presume that it was for an intentional and specific reason. Would providing such a hook into the system prose a significant threat to system security? Situation Report Although knowing whether a memory pointer is valid could be useful in many scenarios, this is my particular situation: I am writing a driver for a new piece of hardware that is to replace an existing piece of hardware that connects to the PC via USB. My mandate is to write the new driver such that calls to the existing API for the current driver will continue to work in the PC applications in which it is used. Thus the only required changes to existing applications is to load the appropriate driver DLL(s) at startup. The problem here is that the existing driver uses a callback to send received serial messages to the application; a pointer to allocated memory containing the message is passed from the driver to the application via the callback. It is then the responsibility of the application to call another driver API to free the memory by passing back the same pointer from the application to the driver. In this scenario the second API has no way to determine if the application has actually passed back a pointer to valid memory.

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