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  • Oracle Database 12c: Oracle Multitenant Option

    - by hamsun
    1. Why ? 2. What is it ? 3. How ? 1. Why ? The main idea of the 'grid' is to share resources, to make better use of storage, CPU and memory. If a database administrator wishes to implement this idea, he or she must consolidate many databases to one database. One of the concerns of running many applications together in one database is: ‚what will happen, if one of the applications must be restored because of a human error?‘ Tablespace point in time recovery can be used for this purpose, but there are a few prerequisites. Most importantly the tablespaces are strictly separated for each application. Another reason for creating separated databases is security: each customer has his own database. Therefore, there is often a proliferation of smaller databases. Each of them must be maintained, upgraded, each allocates virtual memory and runs background processes thereby wasting resources. Oracle 12c offers another possibility for virtualization, providing isolation at the database level: the multitenant container database holding pluggable databases. 2. What ? Pluggable databases are logical units inside a multitenant container database, which consists of one multitenant container database and up to 252 pluggable databases. The SGA is shared as are the background processes. The multitenant container database holds metadata information common for pluggable databases inside the System and the Sysaux tablespace, and there is just one Undo tablespace. The pluggable databases have smaller System and Sysaux tablespaces, containing just their 'personal' metadata. New data dictionary views will make the information available either on pdb (dba_views) or container level (cdb_views). There are local users, which are known in specific pluggable databases and common users known in all containers. Pluggable databases can be easily plugged to another multitenant container database and converted from a non-CDB. They can undergo point in time recovery. 3. How ? Creating a multitenant container database can be done using the database configuration assistant: There you find the new option: Create as Container Database. If you prefer ‚hand made‘ databases you can execute the command from a instance in nomount state: CREATE DATABASE cdb1 ENABLE PLUGGABLE DATABASE …. And of course this can also be achieved through Enterprise Manager Cloud. A freshly created multitenant container database consists of two containers: the root container as the 'rack' and a seed container, a template for future pluggable databases. There are 4 ways to create other pluggable databases: 1. Create an empty pdb from seed 2. Plug in a non-CDB 3. Move a pdb from another pdb 4. Copy a pdb from another pdb We will discuss option2: how to plug in a non_CDB into a multitenant container database. Three different methods are available : 1. Create an empty pdb and use Datapump in traditional export/import mode or with Transportable Tablespace or Database mode. This method is suitable for pre 12c databases. 2. Create an empty pdb and use GoldenGate replication. When the pdb catches up with the non-CDB, you fail over to the pdb. 3. Databases of Version 12c or higher can be plugged in with the help of the new dbms_pdb Package. This is a demonstration for method 3: Step1: Connect to the non-CDB to be plugged in and create an xml File with description of the database. The xml file is written to $ORACLE_HOME/dbs per default and contains mainly information about the datafiles. Step 2: Check if the non-CDB is pluggable in the multitenant container database: Step 3: Create the pluggable database, connected to the Multitenant container database. With nocopy option the files will be reused, but the tempfile is created anew: A service is created and registered automatically with the listener: Step 4: Delete unnecessary metadata from PDB SYSTEM tablespace: To connect to newly created pdb, edit tnsnames.ora and add entry for new pdb. Connect to plugged-in non_CDB and clean up Data Dictionary to remove entries now maintained in multitenant container database. As all kept objects have to be recompiled it will take a few minutes. Step 5: The plugged-in database will be automatically synchronised by creating common users and roles when opened the first time in read write mode. Step 6: Verify tablespaces and users: There is only one local tablespace (users) and one local user (scott) in the plugged-in non_CDB pdb_orcl. This method of creating plugged_in non_CDB from is fast and easy for 12c databases. The method for deplugging a pluggable database from a CDB is to create a new non_CDB and use the the new full transportable feature of Datapump and drop the pluggable database. About the Author: Gerlinde has been working for Oracle University Germany as one of our Principal Instructors for over 14 years. She started with Oracle 7 and became an Oracle Certified Master for Oracle 10g and 11c. She is a specialist in Database Core Technologies, with profound knowledge in Backup & Recovery, Performance Tuning for DBAs and Application Developers, Datawarehouse Administration, Data Guard and Real Application Clusters.

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  • SQL SERVER – How to Compare the Schema of Two Databases with Schema Compare

    - by Pinal Dave
    Earlier I wrote about An Efficiency Tool to Compare and Synchronize SQL Server Databases and it was very much well received. Since the blog post I have received quite a many question that just like data how we can also compare schema and synchronize it. If you think about comparing the schema manually, it is almost impossible to do so. Table Schema has been just one of the concept but if you really want the all the schema of the database (triggers, views, stored procedure and everything else) it is just impossible task. If you are developer or database administrator who works in the production environment than you know that there are so many different occasions when we have to compare schema of the database. Before deploying any changes to the production server, I personally like to make note of the every single schema change and document it so in case of any issue , I can always go back and refer my documentation. As discussed earlier it is absolutely impossible to do this task without the help of third party tools. I personally use Devart Schema Compare for this task. This is an extremely easy tool. Let us see how it works. First I have two different databases – a) AdventureWorks2012 and b) AdventureWorks2012-V1. There are total three changes between these databases. Here is the list of the same. One of the table has additional column One of the table have new index One of the stored procedure is changed Now let see how dbForge Schema Compare works in this scenario. First open dbForge Schema Compare studio. Click on New Schema Comparison. It will bring you to following screen where we have to configure the database needed to configure. I have selected AdventureWorks2012 and AdventureWorks-V1 databases. In the next screen we can verify various options but for this demonstration we will keep it as it is. We will not change anything in schema mapping screen as in our case it is not required but generically if you are comparing across schema you may need this. This is the most important screen as on this screen we select which kind of object we want to compare. You can see the options which are available to select. The screen lets you select the objects from SQL Server 2000 to SQL Server 2012. Once you click on compare in previous screen it will bring you to this screen, which will essentially display the comparative difference between two of the databases which we had selected in earlier screen. As mentioned above there are three different changes in the database and the same has been listed over here. Two of the changes belongs to the tables and one changes belong to the procedure. Let us click each of them one by one to see what is the difference between them. In very first option we can see that there is an additional column in another database which did not exist earlier. In this example we can see that AdventureWorks2012 database have an additional index. Following example is very interesting as in this case, we have changed the definition of the stored procedure and the result pan contains the same. dbForget Schema Compare very effectively identify the changes in schema and lists them neatly to developers. Here is one more screen. This software not only compares the schema but also provides the options to update or drop them as per the choice. I think this is brilliant option. Well, I have been using schema compare for quite a while and have found it very useful. Here are few of the things which dbForge Schema Compare can do for developers and DBAs. Compare and synchronize SQL Server database schemas Compare schemas of live database and SQL Server backup Generate comparison reports in Excel and HTML formats Eliminate mistakes in schema changes propagation across environments Track production database changes and customizations Automate migration of schema changes using command line interface I suggest that you try out dbForge Schema Compare and let me know what you think of this product. Reference: Pinal Dave (http://blog.sqlauthority.com) Filed under: PostADay, SQL, SQL Authority, SQL Query, SQL Server, SQL Tips and Tricks, SQL Utility, T SQL

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  • How John Got 15x Improvement Without Really Trying

    - by rchrd
    The following article was published on a Sun Microsystems website a number of years ago by John Feo. It is still useful and worth preserving. So I'm republishing it here.  How I Got 15x Improvement Without Really Trying John Feo, Sun Microsystems Taking ten "personal" program codes used in scientific and engineering research, the author was able to get from 2 to 15 times performance improvement easily by applying some simple general optimization techniques. Introduction Scientific research based on computer simulation depends on the simulation for advancement. The research can advance only as fast as the computational codes can execute. The codes' efficiency determines both the rate and quality of results. In the same amount of time, a faster program can generate more results and can carry out a more detailed simulation of physical phenomena than a slower program. Highly optimized programs help science advance quickly and insure that monies supporting scientific research are used as effectively as possible. Scientific computer codes divide into three broad categories: ISV, community, and personal. ISV codes are large, mature production codes developed and sold commercially. The codes improve slowly over time both in methods and capabilities, and they are well tuned for most vendor platforms. Since the codes are mature and complex, there are few opportunities to improve their performance solely through code optimization. Improvements of 10% to 15% are typical. Examples of ISV codes are DYNA3D, Gaussian, and Nastran. Community codes are non-commercial production codes used by a particular research field. Generally, they are developed and distributed by a single academic or research institution with assistance from the community. Most users just run the codes, but some develop new methods and extensions that feed back into the general release. The codes are available on most vendor platforms. Since these codes are younger than ISV codes, there are more opportunities to optimize the source code. Improvements of 50% are not unusual. Examples of community codes are AMBER, CHARM, BLAST, and FASTA. Personal codes are those written by single users or small research groups for their own use. These codes are not distributed, but may be passed from professor-to-student or student-to-student over several years. They form the primordial ocean of applications from which community and ISV codes emerge. Government research grants pay for the development of most personal codes. This paper reports on the nature and performance of this class of codes. Over the last year, I have looked at over two dozen personal codes from more than a dozen research institutions. The codes cover a variety of scientific fields, including astronomy, atmospheric sciences, bioinformatics, biology, chemistry, geology, and physics. The sources range from a few hundred lines to more than ten thousand lines, and are written in Fortran, Fortran 90, C, and C++. For the most part, the codes are modular, documented, and written in a clear, straightforward manner. They do not use complex language features, advanced data structures, programming tricks, or libraries. I had little trouble understanding what the codes did or how data structures were used. Most came with a makefile. Surprisingly, only one of the applications is parallel. All developers have access to parallel machines, so availability is not an issue. Several tried to parallelize their applications, but stopped after encountering difficulties. Lack of education and a perception that parallelism is difficult prevented most from trying. I parallelized several of the codes using OpenMP, and did not judge any of the codes as difficult to parallelize. Even more surprising than the lack of parallelism is the inefficiency of the codes. I was able to get large improvements in performance in a matter of a few days applying simple optimization techniques. Table 1 lists ten representative codes [names and affiliation are omitted to preserve anonymity]. Improvements on one processor range from 2x to 15.5x with a simple average of 4.75x. I did not use sophisticated performance tools or drill deep into the program's execution character as one would do when tuning ISV or community codes. Using only a profiler and source line timers, I identified inefficient sections of code and improved their performance by inspection. The changes were at a high level. I am sure there is another factor of 2 or 3 in each code, and more if the codes are parallelized. The study’s results show that personal scientific codes are running many times slower than they should and that the problem is pervasive. Computational scientists are not sloppy programmers; however, few are trained in the art of computer programming or code optimization. I found that most have a working knowledge of some programming language and standard software engineering practices; but they do not know, or think about, how to make their programs run faster. They simply do not know the standard techniques used to make codes run faster. In fact, they do not even perceive that such techniques exist. The case studies described in this paper show that applying simple, well known techniques can significantly increase the performance of personal codes. It is important that the scientific community and the Government agencies that support scientific research find ways to better educate academic scientific programmers. The inefficiency of their codes is so bad that it is retarding both the quality and progress of scientific research. # cacheperformance redundantoperations loopstructures performanceimprovement 1 x x 15.5 2 x 2.8 3 x x 2.5 4 x 2.1 5 x x 2.0 6 x 5.0 7 x 5.8 8 x 6.3 9 2.2 10 x x 3.3 Table 1 — Area of improvement and performance gains of 10 codes The remainder of the paper is organized as follows: sections 2, 3, and 4 discuss the three most common sources of inefficiencies in the codes studied. These are cache performance, redundant operations, and loop structures. Each section includes several examples. The last section summaries the work and suggests a possible solution to the issues raised. Optimizing cache performance Commodity microprocessor systems use caches to increase memory bandwidth and reduce memory latencies. Typical latencies from processor to L1, L2, local, and remote memory are 3, 10, 50, and 200 cycles, respectively. Moreover, bandwidth falls off dramatically as memory distances increase. Programs that do not use cache effectively run many times slower than programs that do. When optimizing for cache, the biggest performance gains are achieved by accessing data in cache order and reusing data to amortize the overhead of cache misses. Secondary considerations are prefetching, associativity, and replacement; however, the understanding and analysis required to optimize for the latter are probably beyond the capabilities of the non-expert. Much can be gained simply by accessing data in the correct order and maximizing data reuse. 6 out of the 10 codes studied here benefited from such high level optimizations. Array Accesses The most important cache optimization is the most basic: accessing Fortran array elements in column order and C array elements in row order. Four of the ten codes—1, 2, 4, and 10—got it wrong. Compilers will restructure nested loops to optimize cache performance, but may not do so if the loop structure is too complex, or the loop body includes conditionals, complex addressing, or function calls. In code 1, the compiler failed to invert a key loop because of complex addressing do I = 0, 1010, delta_x IM = I - delta_x IP = I + delta_x do J = 5, 995, delta_x JM = J - delta_x JP = J + delta_x T1 = CA1(IP, J) + CA1(I, JP) T2 = CA1(IM, J) + CA1(I, JM) S1 = T1 + T2 - 4 * CA1(I, J) CA(I, J) = CA1(I, J) + D * S1 end do end do In code 2, the culprit is conditionals do I = 1, N do J = 1, N If (IFLAG(I,J) .EQ. 0) then T1 = Value(I, J-1) T2 = Value(I-1, J) T3 = Value(I, J) T4 = Value(I+1, J) T5 = Value(I, J+1) Value(I,J) = 0.25 * (T1 + T2 + T5 + T4) Delta = ABS(T3 - Value(I,J)) If (Delta .GT. MaxDelta) MaxDelta = Delta endif enddo enddo I fixed both programs by inverting the loops by hand. Code 10 has three-dimensional arrays and triply nested loops. The structure of the most computationally intensive loops is too complex to invert automatically or by hand. The only practical solution is to transpose the arrays so that the dimension accessed by the innermost loop is in cache order. The arrays can be transposed at construction or prior to entering a computationally intensive section of code. The former requires all array references to be modified, while the latter is cost effective only if the cost of the transpose is amortized over many accesses. I used the second approach to optimize code 10. Code 5 has four-dimensional arrays and loops are nested four deep. For all of the reasons cited above the compiler is not able to restructure three key loops. Assume C arrays and let the four dimensions of the arrays be i, j, k, and l. In the original code, the index structure of the three loops is L1: for i L2: for i L3: for i for l for l for j for k for j for k for j for k for l So only L3 accesses array elements in cache order. L1 is a very complex loop—much too complex to invert. I brought the loop into cache alignment by transposing the second and fourth dimensions of the arrays. Since the code uses a macro to compute all array indexes, I effected the transpose at construction and changed the macro appropriately. The dimensions of the new arrays are now: i, l, k, and j. L3 is a simple loop and easily inverted. L2 has a loop-carried scalar dependence in k. By promoting the scalar name that carries the dependence to an array, I was able to invert the third and fourth subloops aligning the loop with cache. Code 5 is by far the most difficult of the four codes to optimize for array accesses; but the knowledge required to fix the problems is no more than that required for the other codes. I would judge this code at the limits of, but not beyond, the capabilities of appropriately trained computational scientists. Array Strides When a cache miss occurs, a line (64 bytes) rather than just one word is loaded into the cache. If data is accessed stride 1, than the cost of the miss is amortized over 8 words. Any stride other than one reduces the cost savings. Two of the ten codes studied suffered from non-unit strides. The codes represent two important classes of "strided" codes. Code 1 employs a multi-grid algorithm to reduce time to convergence. The grids are every tenth, fifth, second, and unit element. Since time to convergence is inversely proportional to the distance between elements, coarse grids converge quickly providing good starting values for finer grids. The better starting values further reduce the time to convergence. The downside is that grids of every nth element, n > 1, introduce non-unit strides into the computation. In the original code, much of the savings of the multi-grid algorithm were lost due to this problem. I eliminated the problem by compressing (copying) coarse grids into continuous memory, and rewriting the computation as a function of the compressed grid. On convergence, I copied the final values of the compressed grid back to the original grid. The savings gained from unit stride access of the compressed grid more than paid for the cost of copying. Using compressed grids, the loop from code 1 included in the previous section becomes do j = 1, GZ do i = 1, GZ T1 = CA(i+0, j-1) + CA(i-1, j+0) T4 = CA1(i+1, j+0) + CA1(i+0, j+1) S1 = T1 + T4 - 4 * CA1(i+0, j+0) CA(i+0, j+0) = CA1(i+0, j+0) + DD * S1 enddo enddo where CA and CA1 are compressed arrays of size GZ. Code 7 traverses a list of objects selecting objects for later processing. The labels of the selected objects are stored in an array. The selection step has unit stride, but the processing steps have irregular stride. A fix is to save the parameters of the selected objects in temporary arrays as they are selected, and pass the temporary arrays to the processing functions. The fix is practical if the same parameters are used in selection as in processing, or if processing comprises a series of distinct steps which use overlapping subsets of the parameters. Both conditions are true for code 7, so I achieved significant improvement by copying parameters to temporary arrays during selection. Data reuse In the previous sections, we optimized for spatial locality. It is also important to optimize for temporal locality. Once read, a datum should be used as much as possible before it is forced from cache. Loop fusion and loop unrolling are two techniques that increase temporal locality. Unfortunately, both techniques increase register pressure—as loop bodies become larger, the number of registers required to hold temporary values grows. Once register spilling occurs, any gains evaporate quickly. For multiprocessors with small register sets or small caches, the sweet spot can be very small. In the ten codes presented here, I found no opportunities for loop fusion and only two opportunities for loop unrolling (codes 1 and 3). In code 1, unrolling the outer and inner loop one iteration increases the number of result values computed by the loop body from 1 to 4, do J = 1, GZ-2, 2 do I = 1, GZ-2, 2 T1 = CA1(i+0, j-1) + CA1(i-1, j+0) T2 = CA1(i+1, j-1) + CA1(i+0, j+0) T3 = CA1(i+0, j+0) + CA1(i-1, j+1) T4 = CA1(i+1, j+0) + CA1(i+0, j+1) T5 = CA1(i+2, j+0) + CA1(i+1, j+1) T6 = CA1(i+1, j+1) + CA1(i+0, j+2) T7 = CA1(i+2, j+1) + CA1(i+1, j+2) S1 = T1 + T4 - 4 * CA1(i+0, j+0) S2 = T2 + T5 - 4 * CA1(i+1, j+0) S3 = T3 + T6 - 4 * CA1(i+0, j+1) S4 = T4 + T7 - 4 * CA1(i+1, j+1) CA(i+0, j+0) = CA1(i+0, j+0) + DD * S1 CA(i+1, j+0) = CA1(i+1, j+0) + DD * S2 CA(i+0, j+1) = CA1(i+0, j+1) + DD * S3 CA(i+1, j+1) = CA1(i+1, j+1) + DD * S4 enddo enddo The loop body executes 12 reads, whereas as the rolled loop shown in the previous section executes 20 reads to compute the same four values. In code 3, two loops are unrolled 8 times and one loop is unrolled 4 times. Here is the before for (k = 0; k < NK[u]; k++) { sum = 0.0; for (y = 0; y < NY; y++) { sum += W[y][u][k] * delta[y]; } backprop[i++]=sum; } and after code for (k = 0; k < KK - 8; k+=8) { sum0 = 0.0; sum1 = 0.0; sum2 = 0.0; sum3 = 0.0; sum4 = 0.0; sum5 = 0.0; sum6 = 0.0; sum7 = 0.0; for (y = 0; y < NY; y++) { sum0 += W[y][0][k+0] * delta[y]; sum1 += W[y][0][k+1] * delta[y]; sum2 += W[y][0][k+2] * delta[y]; sum3 += W[y][0][k+3] * delta[y]; sum4 += W[y][0][k+4] * delta[y]; sum5 += W[y][0][k+5] * delta[y]; sum6 += W[y][0][k+6] * delta[y]; sum7 += W[y][0][k+7] * delta[y]; } backprop[k+0] = sum0; backprop[k+1] = sum1; backprop[k+2] = sum2; backprop[k+3] = sum3; backprop[k+4] = sum4; backprop[k+5] = sum5; backprop[k+6] = sum6; backprop[k+7] = sum7; } for one of the loops unrolled 8 times. Optimizing for temporal locality is the most difficult optimization considered in this paper. The concepts are not difficult, but the sweet spot is small. Identifying where the program can benefit from loop unrolling or loop fusion is not trivial. Moreover, it takes some effort to get it right. Still, educating scientific programmers about temporal locality and teaching them how to optimize for it will pay dividends. Reducing instruction count Execution time is a function of instruction count. Reduce the count and you usually reduce the time. The best solution is to use a more efficient algorithm; that is, an algorithm whose order of complexity is smaller, that converges quicker, or is more accurate. Optimizing source code without changing the algorithm yields smaller, but still significant, gains. This paper considers only the latter because the intent is to study how much better codes can run if written by programmers schooled in basic code optimization techniques. The ten codes studied benefited from three types of "instruction reducing" optimizations. The two most prevalent were hoisting invariant memory and data operations out of inner loops. The third was eliminating unnecessary data copying. The nature of these inefficiencies is language dependent. Memory operations The semantics of C make it difficult for the compiler to determine all the invariant memory operations in a loop. The problem is particularly acute for loops in functions since the compiler may not know the values of the function's parameters at every call site when compiling the function. Most compilers support pragmas to help resolve ambiguities; however, these pragmas are not comprehensive and there is no standard syntax. To guarantee that invariant memory operations are not executed repetitively, the user has little choice but to hoist the operations by hand. The problem is not as severe in Fortran programs because in the absence of equivalence statements, it is a violation of the language's semantics for two names to share memory. Codes 3 and 5 are C programs. In both cases, the compiler did not hoist all invariant memory operations from inner loops. Consider the following loop from code 3 for (y = 0; y < NY; y++) { i = 0; for (u = 0; u < NU; u++) { for (k = 0; k < NK[u]; k++) { dW[y][u][k] += delta[y] * I1[i++]; } } } Since dW[y][u] can point to the same memory space as delta for one or more values of y and u, assignment to dW[y][u][k] may change the value of delta[y]. In reality, dW and delta do not overlap in memory, so I rewrote the loop as for (y = 0; y < NY; y++) { i = 0; Dy = delta[y]; for (u = 0; u < NU; u++) { for (k = 0; k < NK[u]; k++) { dW[y][u][k] += Dy * I1[i++]; } } } Failure to hoist invariant memory operations may be due to complex address calculations. If the compiler can not determine that the address calculation is invariant, then it can hoist neither the calculation nor the associated memory operations. As noted above, code 5 uses a macro to address four-dimensional arrays #define MAT4D(a,q,i,j,k) (double *)((a)->data + (q)*(a)->strides[0] + (i)*(a)->strides[3] + (j)*(a)->strides[2] + (k)*(a)->strides[1]) The macro is too complex for the compiler to understand and so, it does not identify any subexpressions as loop invariant. The simplest way to eliminate the address calculation from the innermost loop (over i) is to define a0 = MAT4D(a,q,0,j,k) before the loop and then replace all instances of *MAT4D(a,q,i,j,k) in the loop with a0[i] A similar problem appears in code 6, a Fortran program. The key loop in this program is do n1 = 1, nh nx1 = (n1 - 1) / nz + 1 nz1 = n1 - nz * (nx1 - 1) do n2 = 1, nh nx2 = (n2 - 1) / nz + 1 nz2 = n2 - nz * (nx2 - 1) ndx = nx2 - nx1 ndy = nz2 - nz1 gxx = grn(1,ndx,ndy) gyy = grn(2,ndx,ndy) gxy = grn(3,ndx,ndy) balance(n1,1) = balance(n1,1) + (force(n2,1) * gxx + force(n2,2) * gxy) * h1 balance(n1,2) = balance(n1,2) + (force(n2,1) * gxy + force(n2,2) * gyy)*h1 end do end do The programmer has written this loop well—there are no loop invariant operations with respect to n1 and n2. However, the loop resides within an iterative loop over time and the index calculations are independent with respect to time. Trading space for time, I precomputed the index values prior to the entering the time loop and stored the values in two arrays. I then replaced the index calculations with reads of the arrays. Data operations Ways to reduce data operations can appear in many forms. Implementing a more efficient algorithm produces the biggest gains. The closest I came to an algorithm change was in code 4. This code computes the inner product of K-vectors A(i) and B(j), 0 = i < N, 0 = j < M, for most values of i and j. Since the program computes most of the NM possible inner products, it is more efficient to compute all the inner products in one triply-nested loop rather than one at a time when needed. The savings accrue from reading A(i) once for all B(j) vectors and from loop unrolling. for (i = 0; i < N; i+=8) { for (j = 0; j < M; j++) { sum0 = 0.0; sum1 = 0.0; sum2 = 0.0; sum3 = 0.0; sum4 = 0.0; sum5 = 0.0; sum6 = 0.0; sum7 = 0.0; for (k = 0; k < K; k++) { sum0 += A[i+0][k] * B[j][k]; sum1 += A[i+1][k] * B[j][k]; sum2 += A[i+2][k] * B[j][k]; sum3 += A[i+3][k] * B[j][k]; sum4 += A[i+4][k] * B[j][k]; sum5 += A[i+5][k] * B[j][k]; sum6 += A[i+6][k] * B[j][k]; sum7 += A[i+7][k] * B[j][k]; } C[i+0][j] = sum0; C[i+1][j] = sum1; C[i+2][j] = sum2; C[i+3][j] = sum3; C[i+4][j] = sum4; C[i+5][j] = sum5; C[i+6][j] = sum6; C[i+7][j] = sum7; }} This change requires knowledge of a typical run; i.e., that most inner products are computed. The reasons for the change, however, derive from basic optimization concepts. It is the type of change easily made at development time by a knowledgeable programmer. In code 5, we have the data version of the index optimization in code 6. Here a very expensive computation is a function of the loop indices and so cannot be hoisted out of the loop; however, the computation is invariant with respect to an outer iterative loop over time. We can compute its value for each iteration of the computation loop prior to entering the time loop and save the values in an array. The increase in memory required to store the values is small in comparison to the large savings in time. The main loop in Code 8 is doubly nested. The inner loop includes a series of guarded computations; some are a function of the inner loop index but not the outer loop index while others are a function of the outer loop index but not the inner loop index for (j = 0; j < N; j++) { for (i = 0; i < M; i++) { r = i * hrmax; R = A[j]; temp = (PRM[3] == 0.0) ? 1.0 : pow(r, PRM[3]); high = temp * kcoeff * B[j] * PRM[2] * PRM[4]; low = high * PRM[6] * PRM[6] / (1.0 + pow(PRM[4] * PRM[6], 2.0)); kap = (R > PRM[6]) ? high * R * R / (1.0 + pow(PRM[4]*r, 2.0) : low * pow(R/PRM[6], PRM[5]); < rest of loop omitted > }} Note that the value of temp is invariant to j. Thus, we can hoist the computation for temp out of the loop and save its values in an array. for (i = 0; i < M; i++) { r = i * hrmax; TEMP[i] = pow(r, PRM[3]); } [N.B. – the case for PRM[3] = 0 is omitted and will be reintroduced later.] We now hoist out of the inner loop the computations invariant to i. Since the conditional guarding the value of kap is invariant to i, it behooves us to hoist the computation out of the inner loop, thereby executing the guard once rather than M times. The final version of the code is for (j = 0; j < N; j++) { R = rig[j] / 1000.; tmp1 = kcoeff * par[2] * beta[j] * par[4]; tmp2 = 1.0 + (par[4] * par[4] * par[6] * par[6]); tmp3 = 1.0 + (par[4] * par[4] * R * R); tmp4 = par[6] * par[6] / tmp2; tmp5 = R * R / tmp3; tmp6 = pow(R / par[6], par[5]); if ((par[3] == 0.0) && (R > par[6])) { for (i = 1; i <= imax1; i++) KAP[i] = tmp1 * tmp5; } else if ((par[3] == 0.0) && (R <= par[6])) { for (i = 1; i <= imax1; i++) KAP[i] = tmp1 * tmp4 * tmp6; } else if ((par[3] != 0.0) && (R > par[6])) { for (i = 1; i <= imax1; i++) KAP[i] = tmp1 * TEMP[i] * tmp5; } else if ((par[3] != 0.0) && (R <= par[6])) { for (i = 1; i <= imax1; i++) KAP[i] = tmp1 * TEMP[i] * tmp4 * tmp6; } for (i = 0; i < M; i++) { kap = KAP[i]; r = i * hrmax; < rest of loop omitted > } } Maybe not the prettiest piece of code, but certainly much more efficient than the original loop, Copy operations Several programs unnecessarily copy data from one data structure to another. This problem occurs in both Fortran and C programs, although it manifests itself differently in the two languages. Code 1 declares two arrays—one for old values and one for new values. At the end of each iteration, the array of new values is copied to the array of old values to reset the data structures for the next iteration. This problem occurs in Fortran programs not included in this study and in both Fortran 77 and Fortran 90 code. Introducing pointers to the arrays and swapping pointer values is an obvious way to eliminate the copying; but pointers is not a feature that many Fortran programmers know well or are comfortable using. An easy solution not involving pointers is to extend the dimension of the value array by 1 and use the last dimension to differentiate between arrays at different times. For example, if the data space is N x N, declare the array (N, N, 2). Then store the problem’s initial values in (_, _, 2) and define the scalar names new = 2 and old = 1. At the start of each iteration, swap old and new to reset the arrays. The old–new copy problem did not appear in any C program. In programs that had new and old values, the code swapped pointers to reset data structures. Where unnecessary coping did occur is in structure assignment and parameter passing. Structures in C are handled much like scalars. Assignment causes the data space of the right-hand name to be copied to the data space of the left-hand name. Similarly, when a structure is passed to a function, the data space of the actual parameter is copied to the data space of the formal parameter. If the structure is large and the assignment or function call is in an inner loop, then copying costs can grow quite large. While none of the ten programs considered here manifested this problem, it did occur in programs not included in the study. A simple fix is always to refer to structures via pointers. Optimizing loop structures Since scientific programs spend almost all their time in loops, efficient loops are the key to good performance. Conditionals, function calls, little instruction level parallelism, and large numbers of temporary values make it difficult for the compiler to generate tightly packed, highly efficient code. Conditionals and function calls introduce jumps that disrupt code flow. Users should eliminate or isolate conditionls to their own loops as much as possible. Often logical expressions can be substituted for if-then-else statements. For example, code 2 includes the following snippet MaxDelta = 0.0 do J = 1, N do I = 1, M < code omitted > Delta = abs(OldValue ? NewValue) if (Delta > MaxDelta) MaxDelta = Delta enddo enddo if (MaxDelta .gt. 0.001) goto 200 Since the only use of MaxDelta is to control the jump to 200 and all that matters is whether or not it is greater than 0.001, I made MaxDelta a boolean and rewrote the snippet as MaxDelta = .false. do J = 1, N do I = 1, M < code omitted > Delta = abs(OldValue ? NewValue) MaxDelta = MaxDelta .or. (Delta .gt. 0.001) enddo enddo if (MaxDelta) goto 200 thereby, eliminating the conditional expression from the inner loop. A microprocessor can execute many instructions per instruction cycle. Typically, it can execute one or more memory, floating point, integer, and jump operations. To be executed simultaneously, the operations must be independent. Thick loops tend to have more instruction level parallelism than thin loops. Moreover, they reduce memory traffice by maximizing data reuse. Loop unrolling and loop fusion are two techniques to increase the size of loop bodies. Several of the codes studied benefitted from loop unrolling, but none benefitted from loop fusion. This observation is not too surpising since it is the general tendency of programmers to write thick loops. As loops become thicker, the number of temporary values grows, increasing register pressure. If registers spill, then memory traffic increases and code flow is disrupted. A thick loop with many temporary values may execute slower than an equivalent series of thin loops. The biggest gain will be achieved if the thick loop can be split into a series of independent loops eliminating the need to write and read temporary arrays. I found such an occasion in code 10 where I split the loop do i = 1, n do j = 1, m A24(j,i)= S24(j,i) * T24(j,i) + S25(j,i) * U25(j,i) B24(j,i)= S24(j,i) * T25(j,i) + S25(j,i) * U24(j,i) A25(j,i)= S24(j,i) * C24(j,i) + S25(j,i) * V24(j,i) B25(j,i)= S24(j,i) * U25(j,i) + S25(j,i) * V25(j,i) C24(j,i)= S26(j,i) * T26(j,i) + S27(j,i) * U26(j,i) D24(j,i)= S26(j,i) * T27(j,i) + S27(j,i) * V26(j,i) C25(j,i)= S27(j,i) * S28(j,i) + S26(j,i) * U28(j,i) D25(j,i)= S27(j,i) * T28(j,i) + S26(j,i) * V28(j,i) end do end do into two disjoint loops do i = 1, n do j = 1, m A24(j,i)= S24(j,i) * T24(j,i) + S25(j,i) * U25(j,i) B24(j,i)= S24(j,i) * T25(j,i) + S25(j,i) * U24(j,i) A25(j,i)= S24(j,i) * C24(j,i) + S25(j,i) * V24(j,i) B25(j,i)= S24(j,i) * U25(j,i) + S25(j,i) * V25(j,i) end do end do do i = 1, n do j = 1, m C24(j,i)= S26(j,i) * T26(j,i) + S27(j,i) * U26(j,i) D24(j,i)= S26(j,i) * T27(j,i) + S27(j,i) * V26(j,i) C25(j,i)= S27(j,i) * S28(j,i) + S26(j,i) * U28(j,i) D25(j,i)= S27(j,i) * T28(j,i) + S26(j,i) * V28(j,i) end do end do Conclusions Over the course of the last year, I have had the opportunity to work with over two dozen academic scientific programmers at leading research universities. Their research interests span a broad range of scientific fields. Except for two programs that relied almost exclusively on library routines (matrix multiply and fast Fourier transform), I was able to improve significantly the single processor performance of all codes. Improvements range from 2x to 15.5x with a simple average of 4.75x. Changes to the source code were at a very high level. I did not use sophisticated techniques or programming tools to discover inefficiencies or effect the changes. Only one code was parallel despite the availability of parallel systems to all developers. Clearly, we have a problem—personal scientific research codes are highly inefficient and not running parallel. The developers are unaware of simple optimization techniques to make programs run faster. They lack education in the art of code optimization and parallel programming. I do not believe we can fix the problem by publishing additional books or training manuals. To date, the developers in questions have not studied the books or manual available, and are unlikely to do so in the future. Short courses are a possible solution, but I believe they are too concentrated to be much use. The general concepts can be taught in a three or four day course, but that is not enough time for students to practice what they learn and acquire the experience to apply and extend the concepts to their codes. Practice is the key to becoming proficient at optimization. I recommend that graduate students be required to take a semester length course in optimization and parallel programming. We would never give someone access to state-of-the-art scientific equipment costing hundreds of thousands of dollars without first requiring them to demonstrate that they know how to use the equipment. Yet the criterion for time on state-of-the-art supercomputers is at most an interesting project. Requestors are never asked to demonstrate that they know how to use the system, or can use the system effectively. A semester course would teach them the required skills. Government agencies that fund academic scientific research pay for most of the computer systems supporting scientific research as well as the development of most personal scientific codes. These agencies should require graduate schools to offer a course in optimization and parallel programming as a requirement for funding. About the Author John Feo received his Ph.D. in Computer Science from The University of Texas at Austin in 1986. After graduate school, Dr. Feo worked at Lawrence Livermore National Laboratory where he was the Group Leader of the Computer Research Group and principal investigator of the Sisal Language Project. In 1997, Dr. Feo joined Tera Computer Company where he was project manager for the MTA, and oversaw the programming and evaluation of the MTA at the San Diego Supercomputer Center. In 2000, Dr. Feo joined Sun Microsystems as an HPC application specialist. He works with university research groups to optimize and parallelize scientific codes. Dr. Feo has published over two dozen research articles in the areas of parallel parallel programming, parallel programming languages, and application performance.

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  • Patching and PCI Compliance

    - by Joel Weise
    One of my friends and master of the security universe, Darren Moffat, pointed me to Dan Anderson's blog the other day.  Dan went to Toorcon which is a security conference where he went to a talk on security patching titled, "Stop Patching, for Stronger PCI Compliance".  I realize that often times speakers will use a headline grabbing title to create interest in their talk and this one certainly got my attention.  I did not go to the conference and did not see the presentation, so I can only go by what is in the Toorcon agenda summary and on Dan's blog, but the general statement to stop patching for stronger PCI compliance seems a bit misleading to me.  Clearly patching is important to all systems management and should be a part of any organization's security hygiene.  Further, PCI does require the patching of systems to maintain compliance.  So it's important to mention that organizations should not simply stop patching their systems; and I want to believe that was not the speakers intent. So let's look at PCI requirement 6: "Unscrupulous individuals use security vulnerabilities to gain privileged access to systems. Many of these vulnerabilities are fixed by vendor- provided security patches, which must be installed by the entities that manage the systems. All critical systems must have the most recently released, appropriate software patches to protect against exploitation and compromise of cardholder data by malicious individuals and malicious software." Notice the word "appropriate" in the requirement.  This is stated to give organizations some latitude and apply patches that make sense in their environment and that target the vulnerabilities in question.  Haven't we all seen a vulnerability scanner throw a false positive and flag some module and point to a recommended patch, only to realize that the module doesn't exist on our system?  Applying such a patch would obviously not be appropriate.  This does not mean an organization can ignore the fact they need to apply security patches.  It's pretty clear they must.  Of course, organizations have other options in terms of compliance when it comes to patching.  For example, they could remove a system from scope and make sure that system does not process or contain cardholder data.  [This may or may not be a significant undertaking.  I just wanted to point out that there are always options available.] PCI DSS requirement 6.1 also includes the following note: "Note: An organization may consider applying a risk-based approach to prioritize their patch installations. For example, by prioritizing critical infrastructure (for example, public-facing devices and systems, databases) higher than less-critical internal devices, to ensure high-priority systems and devices are addressed within one month, and addressing less critical devices and systems within three months." Notice there is no mention to stop patching one's systems.  And the note also states organization may apply a risk based approach. [A smart approach but also not mandated].  Such a risk based approach is not intended to remove the requirement to patch one's systems.  It is meant, as stated, to allow one to prioritize their patch installations.   So what does this mean to an organization that must comply with PCI DSS and maintain some sanity around their patch management and overall operational readiness?  I for one like to think that most organizations take a common sense and balanced approach to their business and security posture.  If patching is becoming an unbearable task, review why that is the case and possibly look for means to improve operational efficiencies; but also recognize that security is important to maintaining the availability and integrity of one's systems.  Likewise, whether we like it or not, the cyber-world we live in is getting more complex and threatening - and I dont think it's going to get better any time soon.

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  • Measuring Code Quality

    - by DotNetBlues
    Several months back, I was tasked with measuring the quality of code in my organization. Foolishly, I said, "No problem." I figured that Visual Studio has a built-in code metrics tool (Analyze -> Calculate Code Metrics) and that would be a fine place to start with. I was right, but also very wrong. The Visual Studio calculates five primary metrics: Maintainability Index, Cyclomatic Complexity, Depth of Inheritance, Class Coupling, and Lines of Code. The first two are figured at the method level, the second at (primarily) the class level, and the last is a simple count. The first question any reasonable person should ask is "Which one do I look at first?" The first question any manager is going to ask is, "What one number tells me about the whole application?" My answer to both, in a way, was "Maintainability Index." Why? Because each of the other numbers represent one element of quality while MI is a composite number that includes Cyclomatic Complexity. I'd be lying if I said no consideration was given to the fact that it was abstract enough that it's harder for some surly developer (I've been known to resemble that remark) to start arguing why a high coupling or inheritance is no big deal or how complex requirements are to blame for complex code. I should also note that I don't think there is one magic bullet metric that will tell you objectively how good a code base is. There are a ton of different metrics out there, and each one was created for a specific purpose in mind and has a pet theory behind it. When you've got a group of developers who aren't accustomed to measuring code quality, picking a 0-100 scale, non-controversial metric that can be easily generated by tools you already own really isn't a bad place to start. That sort of answers the question a developer would ask, but what about the management question; how do you dashboard this stuff when Visual Studio doesn't roll up the numbers to the solution level? Since VS does roll up the MI to the project level, I thought I could just figure out what sort of weighting Microsoft used to roll method scores up to the class level and then to the namespace and project levels. I was a bit surprised by the answer: there is no weighting. That means that a class with one 1300 line method (which will score a 0 MI) and one empty constructor (which will score a 100 MI) will have an overall MI of a respectable 50. Throw in a couple of DTOs that are nothing more than getters and setters (which tend to score 95 or better) and the project ends up looking really, really healthy. The next poor bastard who has to work on the application is probably not going to be singing the praises of its maintainability, though. For the record, that 1300 line method isn't a hypothetical, either. So, what does one do with that? Well, I decided to weight the average by the Lines of Code per method. For our above example, the formula for the class's MI becomes ((1300 * 0) + (1 * 100))/1301 = .077, rounded to 0. Sounds about right. Continue the pattern for namespace, project, solution, and even multi-solution application MI scores. This can be done relatively easily by using the "export to Excel" button and running a quick formula against the data. On the short list of follow-up questions would be, "How do I improve my application's score?" That's an answer for another time, though.

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  • What is the right way to process inconsistent data files?

    - by Tahabi
    I'm working at a company that uses Excel files to store product data, specifically, test results from products before they are shipped out. There are a few thousand spreadsheets with anywhere from 50-100 relevant data points per file. Over the years, the schema for the spreadsheets has changed significantly, but not unidirectionally - in the sense that, changes often get reverted and then re-added in the space of a few dozen to few hundred files. My project is to convert about 8000 of these spreadsheets into a database that can be queried. I'm using MongoDB to deal with the inconsistency in the data, and Python. My question is, what is the "right" or canonical way to deal with the huge variance in my source files? I've written a data structure which stores the data I want for the latest template, which will be the final template used going forward, but that only helps for a few hundred files historically. Brute-forcing a solution would mean writing similar data structures for each version/template - which means potentially writing hundreds of schemas with dozens of fields each. This seems very inefficient, especially when sometimes a change in the template is as little as moving a single line of data one row down or splitting what used to be one data field into two data fields. A slightly more elegant solution I have in mind would be writing schemas for all the variants I can find for pre-defined groups in the source files, and then writing a function to match a particular series of files with a series of variants that matches that set of files. This is because, more often that not, most of the file will remain consistent over a long period, only marred by one or two errant sections, but inside the period, which section is inconsistent, is inconsistent. For example, say a file has four sections with three data fields, which is represented by four Python dictionaries with three keys each. For files 7000-7250, sections 1-3 will be consistent, but section 4 will be shifted one row down. For files 7251-7500, 1-3 are consistent, section 4 is one row down, but a section five appears. For files 7501-7635, sections 1 and 3 will be consistent, but section 2 will have five data fields instead of three, section five disappears, and section 4 is still shifted down one row. For files 7636-7800, section 1 is consistent, section 4 gets shifted back up, section 2 returns to three cells, but section 3 is removed entirely. Files 7800-8000 have everything in order. The proposed function would take the file number and match it to a dictionary representing the data mappings for different variants of each section. For example, a section_four_variants dictionary might have two members, one for the shifted-down version, and one for the normal version, a section_two_variants might have three and five field members, etc. The script would then read the matchings, load the correct mapping, extract the data, and insert it into the database. Is this an accepted/right way to go about solving this problem? Should I structure things differently? I don't know what to search Google for either to see what other solutions might be, though I believe the problem lies in the domain of ETL processing. I also have no formal CS training aside from what I've taught myself over the years. If this is not the right forum for this question, please tell me where to move it, if at all. Any help is most appreciated. Thank you.

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  • Must-see sessions at TCUK11

    - by Roger Hart
    Technical Communication UK is probably the best professional conference I've been to. Last year, I spoke there on content strategy, and this year I'll be co-hosting a workshop on embedded user assistance. Obviously, I'd love people to come along to that; but there are some other sessions I'd like to flag up for anybody thinking of attending. Tuesday 20th Sept - workshops This will be my first year at the pre-conference workshop day, and I'm massively glad that our workshop hasn't been scheduled along-side the one I'm really interested in. My picks: It looks like you're embedding user assistance. Would you like help? My colleague Dom and I are presenting this one. It's our paen to Clippy, to the brilliant idea he represented, and the crashing failure he was. Less precociously, we'll be teaching embedded user assistance, Red Gate style. Statistics without maths: acquiring, visualising and interpreting your data This doesn't need to do anything apart from what it says on the tin in order to be gold dust. But given the speakers, I suspect it will. A data-informed approach is a great asset to technical communications, so I'd recommend this session to anybody event faintly interested. The speakers here have a great track record of giving practical, accessible introductions to big topics. Go along. Wednesday 21st Sept - day one There's no real need to recommend the keynote for a conference, but I will just point out that this year it's Google's Patrick Hofmann. That's cool. You know what else is cool: Focus on the user, the rest follows An intro to modelling customer experience. This is a really exciting area for tech comms, and potentially touches on one of my personal hobby-horses: the convergence of technical communication and marketing. It's all part of delivering customer experience, and knowing what your users need lets you help them, sell to them, and delight them. Content strategy year 1: a tale from the trenches It's often been observed that content strategy is great at banging its own drum, but not so hot on compelling case studies. Here you go, folks. This is the presentation I'm most excited about so far. On a mission to communicate! Skype help their users communicate, but how do they communicate with them? I guess we'll find out. Then there's the stuff that I'm not too excited by, but you might just be. The standards geeks and agile freaks can get together in a presentation on the forthcoming ISO standards for agile authoring. Plus, there's a session on VBA for tech comms. I do have one gripe about day 1. The other big UK tech comms conference, UA Europe, have - I think - netted the more interesting presentation from Ellis Pratt. While I have no doubt that his TCUK case study on producing risk assessments will be useful, I'd far rather go to his talk on game theory for tech comms. Hopefully UA Europe will record it. Thursday 22nd Sept - day two Day two has a couple of slots yet to be confirmed. The rumour is that one of them will be the brilliant "Questions and rants" session from last year. I hope so. It's not ranting, but I'll be going to: RTFMobile: beyond stating the obvious Ultan O'Broin is an engaging speaker with a lot to say, and mobile is one of the most interesting and challenging new areas for tech comms. Even if this weren't a research-based presentation from a company with buckets of technology experience, I'd be going. It is, and you should too. Pattern recognition for technical communicators One of the best things about TCUK is the tendency to include sessions that tackle the theoretical and bring them towards the practical. Kai and Chris delivered cracking and well-received talks last year, and I'm looking forward to seeing what they've got for us on some of the conceptual underpinning of technical communication. Developing an interactive non-text learning programme Annoyingly, this clashes with Pattern Recognition, so I hope at least one of the streams is recorded again this year. The idea of communicating complex information without words us fascinating and this sounds like a great example of this year's third stream: "anything but text". For the localization and DITA crowds, there's rich pickings on day two, though I'm not sure how many of those sessions I'm interested in. In the 13:00 - 13:40 slot, there's an interesting clash between Linda Urban on re-use and training content, and a piece on minimalism I'm sorely tempted by. That's my pick of #TCUK11. I'll be doing a round-up blog after the event, and probably talking a bit more about it beforehand. I'm also reliably assured that there are still plenty of tickets.

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  • Design Pattern for Complex Data Modeling

    - by Aaron Hayman
    I'm developing a program that has a SQL database as a backing store. As a very broad description, the program itself allows a user to generate records in any number of user-defined tables and make connections between them. As for specs: Any record generated must be able to be connected to any other record in any other user table (excluding itself...the record, not the table). These "connections" are directional, and the list of connections a record has is user ordered. Moreover, a record must "know" of connections made from it to others as well as connections made to it from others. The connections are kind of the point of this program, so there is a strong possibility that the number of connections made is very high, especially if the user is using the software as intended. A record's field can also include aggregate information from it's connections (like obtaining average, sum, etc) that must be updated on change from another record it's connected to. To conserve memory, only relevant information must be loaded at any one time (can't load the entire database in memory at load and go from there). I cannot assume the backing store is local. Right now it is, but eventually this program will include syncing to a remote db. Neither the user tables, connections or records are known at design time as they are user generated. I've spent a lot of time trying to figure out how to design the backing store and the object model to best fit these specs. In my first design attempt on this, I had one object managing all a table's records and connections. I attempted this first because it kept the memory footprint smaller (records and connections were simple dicts), but maintaining aggregate and link information between tables became....onerous (ie...a huge spaghettified mess). Tracing dependencies using this method almost became impossible. Instead, I've settled on a distributed graph model where each record and connection is 'aware' of what's around it by managing it own data and connections to other records. Doing this increases my memory footprint but also let me create a faulting system so connections/records aren't loaded into memory until they're needed. It's also much easier to code: trace dependencies, eliminate cycling recursive updates, etc. My biggest problem is storing/loading the connections. I'm not happy with any of my current solutions/ideas so I wanted to ask and see if anybody else has any ideas of how this should be structured. Connections are fairly simple. They contain: fromRecordID, fromTableID, fromRecordOrder, toRecordID, toTableID, toRecordOrder. Here's what I've come up with so far: Store all the connections in one big table. If I do this, either I load all connections at once (one big db call) or make a call every time a user table is loaded. The big issue here: the size of the connections table has the potential to be huge, and I'm afraid it would slow things down. Store in separate tables all the outgoing connections for each user table. This is probably the worst idea I've had. Now my connections are 'spread out' over multiple tables (one for each user table), which means I have to make a separate DB called to each table (or make a huge join) just to find all the incoming connections for a particular user table. I've avoided making "one big ass table", but I'm not sure the cost is worth it. Store in separate tables all outgoing AND incoming connections for each user table (using a flag to distinguish between incoming vs outgoing). This is the idea I'm leaning towards, but it will essentially double the total DB storage for all the connections (as each connection will be stored in two tables). It also means I have to make sure connection information is kept in sync in both places. This is obviously not ideal but it does mean that when I load a user table, I only need to load one 'connection' table and have all the information I need. This also presents a separate problem, that of connection object creation. Since each user table has a list of all connections, there are two opportunities for a connection object to be made. However, connections objects (designed to facilitate communication between records) should only be created once. This means I'll have to devise a common caching/factory object to make sure only one connection object is made per connection. Does anybody have any ideas of a better way to do this? Once I've committed to a particular design pattern I'm pretty much stuck with it, so I want to make sure I've come up with the best one possible.

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  • How would you gather client's data on Google App Engine without using Datastore/Backend Instances too much?

    - by ruslan
    I'm relatively new to StackExchange and not sure if it's appropriate place to ask design question. Site gives me a hint "The question you're asking appears subjective and is likely to be closed". Please let me know. Anyway.. One of the projects I'm working on is online survey engine. It's my first big commercial project on Google App Engine. I need your advice on how to collect stats and efficiently record them in DataStore without bankrupting me. Initial requirements are: After user finishes survey client sends list of pairs [ID (int) + PercentHit (double)]. This list shows how close answers of this user match predefined answers of reference answerers (which identified by IDs). I call them "target IDs". Creator of the survey wants to see aggregated % for given IDs for last hour, particular timeframe or from the beginning of the survey. Some surveys may have thousands of target/reference answerers. So I created entity public class HitsStatsDO implements Serializable { @Id transient private Long id; transient private Long version = (long) 0; transient private Long startDate; @Parent transient private Key parent; // fake parent which contains target id @Transient int targetId; private double avgPercent; private long hitCount; } But writing HitsStatsDO for each target from each user would give a lot of data. For instance I had a survey with 3000 targets which was answered by ~4 million people within one week with 300K people taking survey in first day. Even if we assume they were answering it evenly for 24 hours it would give us ~1040 writes/second. Obviously it hits concurrent writes limit of Datastore. I decided I'll collect data for one hour and save that, that's why there are avgPercent and hitCount in HitsStatsDO. GAE instances are stateless so I had to use dynamic backend instance. There I have something like this: // Contains stats for one hour private class Shard { ReadWriteLock lock = new ReentrantReadWriteLock(); Map<Integer, HitsStatsDO> map = new HashMap<Integer, HitsStatsDO>(); // Key is target ID public void saveToDatastore(); public void updateStats(Long startDate, Map<Integer, Double> hits); } and map with shard for current hour and previous hour (which doesn't stay here for long) private HashMap<Long, Shard> shards = new HashMap<Long, Shard>(); // Key is HitsStatsDO.startDate So once per hour I dump Shard for previous hour to Datastore. Plus I have class LifetimeStats which keeps Map<Integer, HitsStatsDO> in memcached where map-key is target ID. Also in my backend shutdown hook method I dump stats for unfinished hour to Datastore. There is only one major issue here - I have only ONE backend instance :) It raises following questions on which I'd like to hear your opinion: Can I do this without using backend instance ? What if one instance is not enough ? How can I split data between multiple dynamic backend instances? It hard because I don't know how many I have because Google creates new one as load increases. I know I can launch exact number of resident backend instances. But how many ? 2, 5, 10 ? What if I have no load at all for a week. Constantly running 10 backend instances is too expensive. What do I do with data from clients while backend instance is dead/restarting? Thank you very much in advance for your thoughts.

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  • yield – Just yet another sexy c# keyword?

    - by George Mamaladze
    yield (see NSDN c# reference) operator came I guess with .NET 2.0 and I my feeling is that it’s not as wide used as it could (or should) be.   I am not going to talk here about necessarity and advantages of using iterator pattern when accessing custom sequences (just google it).   Let’s look at it from the clean code point of view. Let's see if it really helps us to keep our code understandable, reusable and testable.   Let’s say we want to iterate a tree and do something with it’s nodes, for instance calculate a sum of their values. So the most elegant way would be to build a recursive method performing a classic depth traversal returning the sum.           private int CalculateTreeSum(Node top)         {             int sumOfChildNodes = 0;             foreach (Node childNode in top.ChildNodes)             {                 sumOfChildNodes += CalculateTreeSum(childNode);             }             return top.Value + sumOfChildNodes;         }     “Do One Thing” Nevertheless it violates one of the most important rules “Do One Thing”. Our  method CalculateTreeSum does two things at the same time. It travels inside the tree and performs some computation – in this case calculates sum. Doing two things in one method is definitely a bad thing because of several reasons: ·          Understandability: Readability / refactoring ·          Reuseability: when overriding - no chance to override computation without copying iteration code and vice versa. ·          Testability: you are not able to test computation without constructing the tree and you are not able to test correctness of tree iteration.   I want to spend some more words on this last issue. How do you test the method CalculateTreeSum when it contains two in one: computation & iteration? The only chance is to construct a test tree and assert the result of the method call, in our case the sum against our expectation. And if the test fails you do not know wether was the computation algorithm wrong or was that the iteration? At the end to top it all off I tell you: according to Murphy’s Law the iteration will have a bug as well as the calculation. Both bugs in a combination will cause the sum to be accidentally exactly the same you expect and the test will PASS. J   Ok let’s use yield! That’s why it is generally a very good idea not to mix but isolate “things”. Ok let’s use yield!           private int CalculateTreeSumClean(Node top)         {             IEnumerable<Node> treeNodes = GetTreeNodes(top);             return CalculateSum(treeNodes);         }             private int CalculateSum(IEnumerable<Node> nodes)         {             int sumOfNodes = 0;             foreach (Node node in nodes)             {                 sumOfNodes += node.Value;             }             return sumOfNodes;         }           private IEnumerable<Node> GetTreeNodes(Node top)         {             yield return top;             foreach (Node childNode in top.ChildNodes)             {                 foreach (Node currentNode in GetTreeNodes(childNode))                 {                     yield return currentNode;                 }             }         }   Two methods does not know anything about each other. One contains calculation logic another jut the iteration logic. You can relpace the tree iteration algorithm from depth traversal to breath trevaersal or use stack or visitor pattern instead of recursion. This will not influence your calculation logic. And vice versa you can relace the sum with product or do whatever you want with node values, the calculateion algorithm is not aware of beeng working on some tree or graph.  How about not using yield? Now let’s ask the question – what if we do not have yield operator? The brief look at the generated code gives us an answer. The compiler generates a 150 lines long class to implement the iteration logic.       [CompilerGenerated]     private sealed class <GetTreeNodes>d__0 : IEnumerable<Node>, IEnumerable, IEnumerator<Node>, IEnumerator, IDisposable     {         ...        150 Lines of generated code        ...     }   Often we compromise code readability, cleanness, testability, etc. – to reduce number of classes, code lines, keystrokes and mouse clicks. This is the human nature - we are lazy. Knowing and using such a sexy construct like yield, allows us to be lazy, write very few lines of code and at the same time stay clean and do one thing in a method. That's why I generally welcome using staff like that.   Note: The above used recursive depth traversal algorithm is possibly the compact one but not the best one from the performance and memory utilization point of view. It was taken to emphasize on other primary aspects of this post.

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  • Where should you put constants and why?

    - by Tim Meyer
    In our mostly large applications, we usually have a only few locations for constants: One class for GUI and internal contstants (Tab Page titles, Group Box titles, calculation factors, enumerations) One class for database tables and columns (this part is generated code) plus readable names for them (manually assigned) One class for application messages (logging, message boxes etc) The constants are usually separated into different structs in those classes. In our C++ applications, the constants are only defined in the .h file and the values are assigned in the .cpp file. One of the advantages is that all strings etc are in one central place and everybody knows where to find them when something must be changed. This is especially something project managers seem to like as people come and go and this way everybody can change such trivial things without having to dig into the application's structure. Also, you can easily change the title of similar Group Boxes / Tab Pages etc at once. Another aspect is that you can just print that class and give it to a non-programmer who can check if the captions are intuitive, and if messages to the user are too detailed or too confusing etc. However, I see certain disadvantages: Every single class is tightly coupled to the constants classes Adding/Removing/Renaming/Moving a constant requires recompilation of at least 90% of the application (Note: Changing the value doesn't, at least for C++). In one of our C++ projects with 1500 classes, this means around 7 minutes of compilation time (using precompiled headers; without them it's around 50 minutes) plus around 10 minutes of linking against certain static libraries. Building a speed optimized release through the Visual Studio Compiler takes up to 3 hours. I don't know if the huge amount of class relations is the source but it might as well be. You get driven into temporarily hard-coding strings straight into code because you want to test something very quickly and don't want to wait 15 minutes just for that test (and probably every subsequent one). Everybody knows what happens to the "I will fix that later"-thoughts. Reusing a class in another project isn't always that easy (mainly due to other tight couplings, but the constants handling doesn't make it easier.) Where would you store constants like that? Also what arguments would you bring in order to convince your project manager that there are better concepts which also comply with the advantages listed above? Feel free to give a C++-specific or independent answer. PS: I know this question is kind of subjective but I honestly don't know of any better place than this site for this kind of question. Update on this project I have news on the compile time thing: Following Caleb's and gbjbaanb's posts, I split my constants file into several other files when I had time. I also eventually split my project into several libraries which was now possible much easier. Compiling this in release mode showed that the auto-generated file which contains the database definitions (table, column names and more - more than 8000 symbols) and builds up certain hashes caused the huge compile times in release mode. Deactivating MSVC's optimizer for the library which contains the DB constants now allowed us to reduce the total compile time of your Project (several applications) in release mode from up to 8 hours to less than one hour! We have yet to find out why MSVC has such a hard time optimizing these files, but for now this change relieves a lot of pressure as we no longer have to rely on nightly builds only. That fact - and other benefits, such as less tight coupling, better reuseability etc - also showed that spending time splitting up the "constants" wasn't such a bad idea after all ;-)

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  • How can I gather client's data on Google App Engine without using Datastore/Backend Instances too much?

    - by ruslan
    One of the projects I'm working on is online survey engine. It's my first big commercial project on Google App Engine. I need your advice on how to collect stats and efficiently record them in DataStore without bankrupting me. Initial requirements are: After user finishes survey client sends list of pairs [ID (int) + PercentHit (double)]. This list shows how close answers of this user match predefined answers of reference answerers (which identified by IDs). I call them "target IDs". Creator of the survey wants to see aggregated % for given IDs for last hour, particular timeframe or from the beginning of the survey. Some surveys may have thousands of target/reference answerers. So I created entity public class HitsStatsDO implements Serializable { @Id transient private Long id; transient private Long version = (long) 0; transient private Long startDate; @Parent transient private Key parent; // fake parent which contains target id @Transient int targetId; private double avgPercent; private long hitCount; } But writing HitsStatsDO for each target from each user would give a lot of data. For instance I had a survey with 3000 targets which was answered by ~4 million people within one week with 300K people taking survey in first day. Even if we assume they were answering it evenly for 24 hours it would give us ~1040 writes/second. Obviously it hits concurrent writes limit of Datastore. I decided I'll collect data for one hour and save that, that's why there are avgPercent and hitCount in HitsStatsDO. GAE instances are stateless so I had to use dynamic backend instance. There I have something like this: // Contains stats for one hour private class Shard { ReadWriteLock lock = new ReentrantReadWriteLock(); Map<Integer, HitsStatsDO> map = new HashMap<Integer, HitsStatsDO>(); // Key is target ID public void saveToDatastore(); public void updateStats(Long startDate, Map<Integer, Double> hits); } and map with shard for current hour and previous hour (which doesn't stay here for long) private HashMap<Long, Shard> shards = new HashMap<Long, Shard>(); // Key is HitsStatsDO.startDate So once per hour I dump Shard for previous hour to Datastore. Plus I have class LifetimeStats which keeps Map<Integer, HitsStatsDO> in memcached where map-key is target ID. Also in my backend shutdown hook method I dump stats for unfinished hour to Datastore. There is only one major issue here - I have only ONE backend instance :) It raises following questions on which I'd like to hear your opinion: Can I do this without using backend instance ? What if one instance is not enough ? How can I split data between multiple dynamic backend instances? It hard because I don't know how many I have because Google creates new one as load increases. I know I can launch exact number of resident backend instances. But how many ? 2, 5, 10 ? What if I have no load at all for a week. Constantly running 10 backend instances is too expensive. What do I do with data from clients while backend instance is dead/restarting?

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  • What do you need to know to be a world-class master software developer? [closed]

    - by glitch
    I wanted to bring up this question to you folks and see what you think, hopefully advise me on the matter: let's say you had 30 years of learning and practicing software development in front of you, how would you dedicate your time so that you'd get the biggest bang for your buck. What would you both learn and work on to be a world-class software developer that would make a large impact on the industry and leave behind a legacy? I think that most great developers end up being both broad generalists and specialists in one-two areas of interest. I'm thinking Bill Joy, John Carmack, Linus Torvalds, K&R and so on. I'm thinking that perhaps one approach would be to break things down by categories and establish a base minimum of "software development" greatness. I'm thinking: Operating Systems: completely internalize the core concepts of OS, perhaps gain a lot of familiarity with an OSS one such as Linux. Anything from memory management to device drivers has to be complete second nature. Programming Languages: this is one of those topics that imho has to be fully grokked even if it might take many years. I don't think there's quite anything like going through the process of developing your own compiler, understanding language design trade-offs and so on. Programming Language Pragmatics is one of my favorite books actually, I think you want to have that internalized back to back, and that's just the start. You could go significantly deeper, but I think it's time well spent, because it's such a crucial building block. As a subset of that, you want to really understand the different programming paradigms out there. Imperative, declarative, logic, functional and so on. Anything from assembly to LISP should be at the very least comfortable to write in. Contexts: I believe one should have experience working in different contexts to truly be able to appreciate the trade-offs that are being made every day. Embedded, web development, mobile development, UX development, distributed, cloud computing and so on. Hardware: I'm somewhat conflicted about this one. I think you want some understanding of computer architecture at a low level, but I feel like the concepts that will truly matter will be slightly higher level, such as CPU caching / memory hierarchy, ILP, and so on. Networking: we live in a completely network-dependent era. Having a good understanding of the OSI model, knowing how the Web works, how HTTP works and so on is pretty much a pre-requisite these days. Distributed systems: once again, everything's distributed these days, it's getting progressively harder to ignore this reality. Slightly related, perhaps add solid understanding of how browsers work to that, since the world seems to be moving so much to interfacing with everything through a browser. Tools: Have a really broad toolset that you're familiar with, one that continuously expands throughout the years. Communication: I think being a great writer, effective communicator and a phenomenal team player is pretty much a prerequisite for a lot of a software developer's greatness. It can't be overstated. Software engineering: understanding the process of building software, team dynamics, the requirements of the business-side, all the pitfalls. You want to deeply understand where what you're writing fits from the market perspective. The better you understand all of this, the more of your work will actually see the daylight. This is really just a starting list, I'm confident that there's a ton of other material that you need to master. As I mentioned, you most likely end up specializing in a bunch of these areas as you go along, but I was trying to come up with a baseline. Any thoughts, suggestions and words of wisdom from the grizzled veterans out there who would like to share their thoughts and experiences with this? I'd really love to know what you think!

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  • SEO/Google: How should I handle multiple countries and domains?

    - by Valorized
    Hello. I'm the webmaster of an online shop based in Austria (Europe). Therefore we registered "example.at". We also own different other domain names like "example-shop.com" and "example.info". Currently all those domains are redirected (301) to the .at one. Still available is: "example.net" and "example.org" (and .ws/.cc), unfortunately not available: .de/.eu The .com is currently owned by one of our partners, the contract ends in 2012 but until then we have no chance to get this one. Recently I read more about geo-targeting and I noticed ONE big deal. The tld ".at" is hardly recognised in Germany (google.de) whereas it is excellently listed in Austria (google.at). As a result of the .at I cannot set the target location manually (or to unlisted). More info: https://www.google.com/support/webmasters/bin/answer.py?answer=62399&hl=en This is a big problem. I looked at Google Analytics and - although Germany is 10x as big as Austria - there are more visits from Austria. So, how should I config the domain in order to get the best results in both, Germany and Austria? I thought of some solutions: First I could stop redirecting the .info. Then there would be a duplicate of the .at one. Moreover, in Webmastertools, I could set the target location of the .info to Germany. As the .at still targets Austria, both would be targeted - however I don't now if google punishes one of them because of the duplicate content? Same as 1. but with .net or .org (I think .info is not a "nice" domain and moreover I think search engines prefer .com, .net or .org to .info). Same as 1. (or 2.) but with a rel="canonical" on the new one (pointing to the .at). Con: I don't think this will improve the situation, because it still tells google that the .at one is more important, like: "if .info points to .at, the target may still be Austria". rel="canonical" on the .at pointing to the new (.info or .net or .org). However I fear that this will have a negative impact on the listing on google.at because: "Hey, the well-known .at is not important anymore, so let's focus on the .info which is not well-known." - Therefore: bad position in search results. Redirect .at to the new (.info or .net or .org) with a 301-Redirect. Con: Might be worse than 4, we might loose Page-Rank (or "the value of the page", because google says that page rank is not important anymore). Moreover this might be even more confusing for the customers. In 3. or 4. customers don't get redirected, they do not see the canonical-meta-tag. So, dear experts, please tell me what the best option would be! Thank you very much for your advice in advance and please excuse the long question. I really appreciate this network! Please note: It's exactly the same content AND language. In Austria we speak German.

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  • Is there a way to tell what the download speed is from a site/server

    - by Memor-X
    i'm looking into which ISP i should go with at the place i'm moving into, one ISP which i have been told good things about has data limits (which when breached will drop your speed to dial-up speed) but multiple memberships which, apart from the cheapest membership, have the same data limits (the cheapest has a 10GB data limit) in their fine print, they say that each different membership has different port speeds, one particular part jumps out at me These speeds are the NBN (National Broadband Network) port speed and not the actual Internet data speed which will vary based on numerous factors including destination you are reaching, your network equipment, network congestion etc. i plan to use the net to download DLC and patch updates for games (particular the insanely large update for the Wii U) and games from Steam (if i find any good one other than this one JRPG) and downloading development resources from free sites like Deposit Files and Mediafire since one membership with a 1000GB data limit is $145 with the port speed being 12Mbps/1Mbps (cheapest) while another with the same limit is $190 with the port speed of 100Mbps/40Mbps (expensive) i am wondering how i can tell what the speed coming from site is since i don't want to be wasting money on speed that makes no difference (unlike memory which i rather have to spare) NOTE: the speeds are for a fiber optic network which where my new place is can only connect via fixed wireless which i may not be able to get with this ISP but if i can get this network then good NOTE 2: most of the resources i get from Deposit Files are always about 200 MB or less, if a resource pack is greater then it's split into multiple archives (like .7z.part) while Mediafire i have to see one bigger than 150MB NOTE 3: one update patch for a PS3 game is close to 4 GB (Disgaea 4) which i need to get access to the DLC and on the weekend i downloaded 5 GB for the Final Fantasy XIV Open Beta for the PS3 which took almost 5 hours

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  • Server Performance

    - by sb12
    I know very little about performance tuning of servers etc... so i thought i'd put this up here as i start some research on it, just to get some direction. I am in the process of migrating from my old server to a new one - both are 64 bit machines. One is a few years old, the other brand new (PowerEdge R410). The old server spec is: 2 cpus, 3.4GHz Pentiums, 8G of RAM, Fedora 11 currently installed The new server spec is: 16 cpus, 3.2 GHz Xeon, 16G of RAM, CentOS 6.2 installed. Also RAID10 is on the new server - no RAID on the old one. Both servers currently have the same database (MySQL) with the same data migrated. I wrote a Perl script that simply steps through each row of a table in the database (about 18000 rows) and updates a value in that row. Every row in the table is updated. Out of curiosity i ran this perl script on both machines, just to see how the new server would perform vs. the old one, and it produced interesting results: The old server was twice as fast as the new one to complete. Looking at the database, both are configured exactly the same (the new one being a dump of the old one...)... Anyone any ideas why this would be given the hardware gap between both? As i said i'm about to start some digging, but thought i'd put this up here to maybe get some good direction.... Many thanks in advance..

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  • Apache subdomain not working

    - by tandu
    I'm running apache on my local machine and I'm trying to create a subdomain, but it's not working. Here is what I have (stripped down): <VirtualHost *:80> DocumentRoot /var/www/one ServerName one.localhost </VirtualHost> <VirtualHost *:80> DocumentRoot /var/www/two ServerName two.localhost </VirtualHost> I recently added one. The two entry has been around for a while, and it still works fine (displays the webpage when I go to two.localhost). In fact, I copied the entire two.localhost entry and simply changed two to one, but it's not working. I have tried each of the following: * `apachectl -k graceful` * `apachectl -k restart` * `/etc/init.d/apache2 restart` * `/etc/init.d/apache2 stop && !#:0 start` Apache will complain if /var/www/one does not exist, so I know it's doing something, but when I visit one.localhost in my browser, the browser complains that nothing is there. I put an index.html file there and also tried going to one.localhost/index.html directly, and the browser still won't fine it. This is very perplexing since the entry I copied from two.localhost is exactly the same .. not only that, but if something were wrong I would expect to get a 500 rather than the browser not being able to find anything. The error_log also has nothing extra.

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  • LCD monitor flicker when connected to a laptop using VGA

    - by Björn Lindqvist
    I have a dual screen setup with two AOC e2450Sw monitors connected to a laptop. The laptop has one HDMI and one VGA output. When one of the monitors is connected using VGA, it flickers or displays static noise. The flickering is fairly subtle and only visible on darker colors. But it is there and noticable and appears like horizontal lines. The problem only appears on the monitor connected to the laptop using the VGA cable. If I swap the monitors, the one connected using VGA is displaying the flicker but not the one connected using HDMI. The simple solution would ofcourse be to connect both monitors using HDMI, but since the laptop only has one VGA and one HDMI out that isn't possible. I've tried tweaking the monitor setting using the OSD menu, but it had little or no effect. Update: After several more trouble shooting hours, it seems the problem is not related to the monitor or VGA cable as the problem persists even if I swap the display with another brand and different cables. So it may be the graphics card? Intel HD Graphics 4000. The laptop is Acer Aspire E1-571.

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  • Multiheaded X.org with a single workspace-pool

    - by blauwblaatje
    I've got an idea for x.org/$randomwindowmanager in combination with a multiheaded setup, but I haven't figured out how it should work. Also I don't really know where to place the feature request. Now for the idea. I've been working with screen (wikipedia:GNU_Screen) for some years now. One thing I like about it, is the fact that I can get a multi-display mode (screen -x), so you can have multiple terminals all connected to the same screen. The fun thing about it, is that you can get 2 terminals with the same content and switch my onscreen layout, without moving the terminals. I admit, in screen it's not extremely useful, but I think for a wm it can be. Imagine this. You've got two monitors and 4 workdesks. On one workdesk I've got my IDE with code, on the second one I've got the output, on the third one I've got the documentation and on the forth one I've got my e-mail and IM clients. At one moment, I want my IDE and output on my monitors, another moment my code and documentation and Yet another moment my IM to consult a colleague and documentation or code. Finally my colleague comes to help me at my desk. I'd like it if we could both watch the same workdesk without him sitting on my lap, so I turn one monitor so he can see it better. It would be great if we could see the same thing that's on my monitor (exclude mousepointer). The thing with most WMs is that your workspaces on the two monitors are either separated or glued together. If they're separated, you can change workspaces on each monitor autonomous, but you can't exchange applications between monitors because they're different x-clients (iirc). If they're glued together (xinerama), you can exchange the applications, but when changing your workspace, the other monitors change too. So, what I'd like to know is this. Is this already possible or should I submit a feature request somewhere (and if so, where?)

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  • Confused with creating an ODBC connection, apparently I have two separate odbcad32.exe files?

    - by Hoser
    Alright, this is my first time working with this so forgive me if I'm a little confusing or vague. I have a server with Windows Server 2008 Standard without Hyper-v (6.0, Build 6002). I'm running a small website off this server and using a Microsoft Access database to store some information coming in through the website. I'm sure the PHP I have written to open the ODBC connection is correct as it has worked for me when I created this website in a testing environment on a laptop. My current issue now is that it seems like I have two different odbcad32.exe's, and one doesn't appear to have a driver for a .accdb file, and only a .mdb file. The other has a driver for both. The first one I speak of has a driver titled 'Driver do Microsoft Access (.mdb)', the second one has a driver titled 'Microsoft Access Driver (.mdb, .accdb)'. I access the first odbcad32.exe by going to C:\Windows\SysWOW64\odbcad32.exe, and then the one that seems to have the driver I need I go to Control Panel-Administrative Tools-Data Sources(ODBC) and simply create a new connection in the System DNS tab. Whenever I make changes to the one that I access through the Control Panel, I see no changes, however if I use the odbcad32.exe file in SysWOW64 I do get some changes in the errors that come back to me. The main difference I noticed is that when I set up an ODBC connection with the Control Panel method it said it simply couldn't find the ODBC connection, but when I made a .mdb connection in the SysWOW64 one (and pointed it to a .accdb file) it says Cannot open database '(unknown)'. It may not be a database that your application recognizes, or the file may be corrupt. Which makes it seem like it is this odbcad32.exe version in SySWOW64 that is being recognized as the 'correct' one. Is there any way to fix this? I've tried to be as thorough as possible but if I've been confusing or left anything out let me know.

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  • Confused about home screen widget size in normal screen and larget screen

    - by kknight
    I am designing a home screen widget. The widget layout file is like below. <RelativeLayout xmlns:android="http://schemas.android.com/apk/res/android" android:id="@+id/widget" android:layout_width="240dip" android:layout_height="200dip" android:background="@drawable/base_all" /> I ran this widget on a HTC Hero device, which has a screen of 320 pixels * 480 pixels with mdpi. It ran perfect on HTC Hero. The widget takes 3 cells * 2 cells space, i.e. 240 pixels * 200 pixels. Then I ran this widget on a Nexus One device, which has a screen of 480 pixels * 800 pixels, mdpi. Since Nexus One also is mdpi, so I though 240dip is equivalent to 240 pixels on Nexus One and 200dip is equivalent to 200 pixels on Nexus One, so the widget will not take 3 cells * 2 cells space on Nexus One device. To my surprise, when running on Nexus One device, the widget take exact 3 cells * 2 cells, about 360 pixels * 300 pixels, on Nexus One device. I am confused. The layout xml above specifies 240dip in width and 200dip in height for the widget, but why did it take 360 pixels * 300 pixels on Nexus One Device? What am I missing? Thanks.

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  • How to get list of files which are currently being diffed in vim

    - by Yogesh Arora
    I am writing a vim plugin in which i need to determine all those files which are currently being diffed. That is the ones for which diff is set. I have been going through the manual but could not find much. Is it possible to do this. This question is actually related to question how-to-detect-the-position-of-window-in-vim. In that question i was trying to get the position of window, so as to detect which one of the diffs is the right one and which is left one. The solution i got was to use winnr() That solution can work only if there are only 2 windows(the ones being diffed). I want to make it generic so that even if multiple windows are open in vim, i can determine which one is on left and which one is right. This is what i was thinking to solve the problem Get a list of all listed buffers For each of this buffers determine if diff is 1 for that If diff is 1 use bufwinnr() to gets it window number. From the window numbers determine which one is left and which one is right. left one will have smaller window number And then determine if current buffer(in which alt-left`alt-right` is pressed) is left or right using winnr of current buffer. Now the pieces that are missing are 1 and 2. For 1 ls can be used but i need to parse its output. Is there a straightfwd way to get list of all listed buffers. And then is there a way to check if for that buffer diff is 1 or not. Any suggestions for a simpler solution are also appreciated.

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  • How should I design my MYSQL table/s?

    - by yaya3
    I built a really basic php/mysql site for an architect that uses one 'projects' table. The website showcases various projects that he has worked on. Each project contained one piece of text and one series of images. Original projects table (create syntax): CREATE TABLE `projects` ( `project_id` int(11) NOT NULL auto_increment, `project_name` text, `project_text` text, `image_filenames` text, `image_folder` text, `project_pdf` text, PRIMARY KEY (`project_id`) ) ENGINE=MyISAM AUTO_INCREMENT=8 DEFAULT CHARSET=latin1; The client now requires the following, and I'm not sure how to handle the expansions in my DB. My suspicion is that I will need an additional table. Each project now have 'pages'. Pages either contain... One image One "piece" of text One image and one piece of text. Each page could use one of three layouts. As each project does not currently have more than 4 pieces of text (a very risky assumption) I have expanded the original table to accommodate everything. New projects table attempt (create syntax): CREATE TABLE `projects` ( `project_id` int(11) NOT NULL AUTO_INCREMENT, `project_name` text, `project_pdf` text, `project_image_folder` text, `project_img_filenames` text, `pages_with_text` text, `pages_without_img` text, `pages_layout_type` text, `pages_title` text, `page_text_a` text, `page_text_b` text, `page_text_c` text, `page_text_d` text, PRIMARY KEY (`project_id`) ) ENGINE=MyISAM AUTO_INCREMENT=8 DEFAULT CHARSET=latin1; In trying to learn more about MYSQL table structuring I have just read an intro to normalization and A Simple Guide to Five Normal Forms in Relational Database Theory. I'm going to keep reading! Thanks in advance

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  • Problem pushing multiple view controllers onto navigation controller stack

    - by Jim
    Hi, I am trying to push three view controllers onto the navigation controller. [self.navigationController pushViewController:one animated:YES]; [self.navigationController pushViewController:two animated:YES]; [self.navigationController pushViewController:three animated:YES]; The desired behavior is that view three will show, and when the back button is pressed it will go to view two and then to view one... What actually happens is that view one is visible and pressing back goes to view two and then back again it goes to view one. Which is to say that view one is shown instead of view three. Very strangely, looking at the viewController array of the navigationController after the calls above show the right entries, and looking at the visibleViewController property shows that it has view three in it... even though view one is visible. If i navigate to a sub view from the visible view one (that shows in the place of view three) and press back from that sub view... it goes to view three. It looks like it is showing view one, but knows it is on view three... I am completely confused... any ideas? Jim

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  • [Ruby] Object assignment and pointers

    - by Jergason
    I am a little confused about object assignment and pointers in Ruby, and coded up this snippet to test my assumptions. class Foo attr_accessor :one, :two def initialize(one, two) @one = one @two = two end end bar = Foo.new(1, 2) beans = bar puts bar puts beans beans.one = 2 puts bar puts beans puts beans.one puts bar.one I had assumed that when I assigned bar to beans, it would create a copy of the object, and modifying one would not affect the other. Alas, the output shows otherwise. ^_^[jergason:~]$ ruby test.rb #<Foo:0x100155c60> #<Foo:0x100155c60> #<Foo:0x100155c60> #<Foo:0x100155c60> 2 2 I believe that the numbers have something to do with the address of the object, and they are the same for both beans and bar, and when I modify beans, bar gets changed as well, which is not what I had expected. It appears that I am only creating a pointer to the object, not a copy of it. What do I need to do to copy the object on assignment, instead of creating a pointer? Tests with the Array class shows some strange behavior as well. foo = [0, 1, 2, 3, 4, 5] baz = foo puts "foo is #{foo}" puts "baz is #{baz}" foo.pop puts "foo is #{foo}" puts "baz is #{baz}" foo += ["a hill of beans is a wonderful thing"] puts "foo is #{foo}" puts "baz is #{baz}" This produces the following wonky output: foo is 012345 baz is 012345 foo is 01234 baz is 01234 foo is 01234a hill of beans is a wonderful thing baz is 01234 This blows my mind. Calling pop on foo affects baz as well, so it isn't a copy, but concatenating something onto foo only affects foo, and not baz. So when am I dealing with the original object, and when am I dealing with a copy? In my own classes, how can I make sure that assignment copies, and doesn't make pointers? Help this confused guy out.

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