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• Dealing with "I-am-cool-and-you-are-dumb" manager [closed]

  - by Software Guy

  I have been working with a software company for about 6 months now. I like the projects I work on there and I really like all the people there except for 1 guy. That guy is technically smart, and he is a co-founder of the company. He is an okay guy in person (the kind you wouldn't want to care about much) but things get tricky when he is your manager. In general I am all okay but there are times when I feel I am not being treated fairly: He doesn't give much thought to when he makes mistakes and when I do something similar, he is super critical. Recently he went as far as to say "I am not sure if I can trust you with this feature". The detais of this specific case are this: I was working on this feature, and I was already a couple of hours over my normal working hours, and then I decided to stop and continue tomorrow. We use git, and I like to commit changes locally and only push when I feel they are ready. This manager insists that I push all the changes to the central repo (in case my hard drive crashes). So I push the change, and the ticket is marked as "to be tested". Next day I come in, he sits next to me and starts complaining and says that I posted above. I really didn't know what to say, I tried to explain to him that the ticket is still being worked upon but he didn't seem to listen. He interrupts me in-between when I am coding, which I do not mind, but when I do that same, his face turns like this :| and reacts as if his work was super important and I am just wasting his time. He asks me to accumulate all questions, and then ask him altogether which is not always possible, as you need a clarification before you can continue on a feature implementation. And when I am coding, he talks on the phone with his customers next to me (when he can go to the meeting room with his laptop) and doesn't care. He made me switch to a whole new IDE (from Netbeans to a commercial IDE costing a lot of money) for a really tiny feature (which I later found out was in Netbeans as well!). I didn't make a big deal out of it as I am equally comfortable working with this new IDE, but I couldn't get the science behind his obsession. He said this feature makes sure that if any method is updated by a programmer, the IDE will turn the method name to red in places where it is used. I told him that I do not have a problem since I always search for method usage in the project and make sure its updated. IDEs even have refactoring features for exactly that, but... I recently implemented a feature for a project, and I was happy about it and considering him a senior, I asked him his comments about the implementation quality.. he thought long and hard, made a few funny faces, and when he couldn't find anything, he said "ummm, your program will crash if JS is disabled" - he was wrong, since I had made sure it would work fine with default values even if JS was disabled. I told him that and then he said "oh okay". BUT, the funny thing is, a few days back, he implemented something and I objected with "But that would not run if JS is disabled" and his response was "We don't have to care about people who disable JS" :-/ Once he asked me to investigate if there was a way to modify a CMS generated menu programmatically by extending the CMS, I did my research and told him that the only was is to inject a menu item using JavaScript / jQuery and his reaction was "ah that's ugly, and hacky, not acceptable" and two days later, I see that feature implemented in the same way as I had suggested. The point is, his reaction was not respectful at all, even if what I proposed was hacky, he should be respectful, that I know what's hacky and if I am suggesting something hacky, there must be a reason for it. There are plenty of other reasons / examples where I feel I am not being treated fairly. I want your advice as to what is it that I am doing wrong and how to deal with such a situation. The other guys in the team are actually very good people, and I do not want to leave the job either (although I could, if I want to). All I want is respect and equal treatment. I have thought about talking to this guy in a face to face meeting, but that worries me that his attitude might get worse and make things more difficult for me (since he doesn't seem to be the guy who thinks he can be wrong too). I am also considering talking to the other co-founder but I am not sure how he will take it (as both founders have been friends forever). Thanks for reading the long message, I really appreciate your help.

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• Training a 'replacement', how to enforce standards?

  - by Mohgeroth

  Not sure that this is the right stack exchange site to ask this of, but here goes... Scope I work for a small company that employs a few hundred people. The development team for the company is small and works out of visual foxpro. A specific department in the company hired me in as a 'lone gunman' to fix and enhance a pre-existing invoicing system. I've successfully taken an Access application that suffered from a lot of risks and limitations and converted it into a C# application driven off of a SQL server backend. I have recently obtained my undergraduate and am no expert by any means. To help make up for that I've felt that earning microsoft certifications will force me to understand more about .net and how it functions. So, after giving my notice with 9 months in advance, 3 months ago a replacement finally showed up. Their role is to learn what I have been designing to an attempt to support the applications designed in C#. The Replacement Fresh out of college with no real-world work experience, the first instinct for anything involving data was and still is listboxes... any time data is mentioned the list box is the control of choice for the replacement. This has gotten to the point, no matter how many times I discuss other controls, where I've seen 5 listboxes on a single form. Classroom experience was almost all C++ console development. So, an example of where I have concern is in a winforms application: Users need to key Reasons into a table to select from later. Given that I know that a strongly typed data set exists, I can just drag the data source from the toolbox and it would create all of this for me. I realize this is a simple example but using databinding is the key. For the past few months now we have been talking about the strongly typed dataset, how to use it and where it interacts with other controls. Data sets, how they work in relation to binding sources, adapters and data grid views. After handing this project off I expected questions about how to implement these since for me this is the way to do it. What happened next simply floors me: An instance of an adapter from the strongly typed dataset was created in the activate event of the form, a table was created and filled with data. Then, a loop was made to manually add rows to a listbox from this table. Finally, a variable was kept to do lookups to figure out what ID the record was for updates if required. How do they modify records you ask? That was my first question too. You won't believe how simple it is, all you do it double click and they type into a pop-up prompt the new value to change it to. As a data entry operator, all the modal popups would drive me absolutely insane. The final solution exceeds 100 lines of code that must be maintained. So my concern is that none of this is sinking in... the department is only allowed 20 hours a week of their time. Up until last week, we've only been given 4-5 hours a week if I'm lucky. The past week or so, I've been lucky to get 10. Question WHAT DO I DO?! I have 4 weeks left until I leave and they fully 'support' this application. I love this job and the opportunity it has given me but it's time for me to spread my wings and find something new. I am in no way, shape or form convinced that they are ready to take over. I do feel that the replacement has the technical ability to 'figure it out' but instead of learning they just write code to do all of this stuff manually. If the replacement wants to code differently in the end, as long as it works I'm fine with that as horrifiying at it looks. However to support what I have designed they MUST to understand how it works and how I have used controls and the framework to make 'magic' happen. This project has about 40 forms, a database with over 30 some odd tables, triggers and stored procedures. It relates labor to invoices to contracts to projections... it's not as simple as it was three years ago when I began this project and the department is now in a position where they cannot survive without it. How in the world can I accomplish any of the following?: Enforce standards or understanding in constent design when the department manager keeps telling them they can do it however they want to Find a way to engage the replacement in active learning of the framework and system design that support must be given for Gracefully inform sr. management that 5-9 hours a week is simply not enough time to learn about the department, pre-existing processes, applications that need to be supported AND determine where potential enhancements to the system go... Yes I know this is a wall of text, thanks for reading through me but I simply don't know what I should be doing. For me, this job is a monster of a reference and things would look extremely bad if I left and things fell apart. How do I handle this?

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• Computer Networks UNISA - Chap 8 &ndash; Wireless Networking

  - by MarkPearl

  After reading this section you should be able to Explain how nodes exchange wireless signals Identify potential obstacles to successful transmission and their repercussions, such as interference and reflection Understand WLAN architecture Specify the characteristics of popular WLAN transmission methods including 802.11 a/b/g/n Install and configure wireless access points and their clients Describe wireless MAN and WAN technologies, including 802.16 and satellite communications The Wireless Spectrum All wireless signals are carried through the air by electromagnetic waves. The wireless spectrum is a continuum of the electromagnetic waves used for data and voice communication. The wireless spectrum falls between 9KHZ and 300 GHZ. Characteristics of Wireless Transmission Antennas Each type of wireless service requires an antenna specifically designed for that service. The service’s specification determine the antenna’s power output, frequency, and radiation pattern. A directional antenna issues wireless signals along a single direction. An omnidirectional antenna issues and receives wireless signals with equal strength and clarity in all directions The geographical area that an antenna or wireless system can reach is known as its range Signal Propagation LOS (line of sight) uses the least amount of energy and results in the reception of the clearest possible signal. When there is an obstacle in the way, the signal may… pass through the object or be obsrobed by the object or may be subject to reflection, diffraction or scattering. Reflection – waves encounter an object and bounces off it. Diffraction – signal splits into secondary waves when it encounters an obstruction Scattering – is the diffusion or the reflection in multiple different directions of a signal Signal Degradation Fading occurs as a signal hits various objects. Because of fading, the strength of the signal that reaches the receiver is lower than the transmitted signal strength. The further a signal moves from its source, the weaker it gets (this is called attenuation) Signals are also affected by noise – the electromagnetic interference) Interference can distort and weaken a wireless signal in the same way that noise distorts and weakens a wired signal. Frequency Ranges Older wireless devices used the 2.4 GHZ band to send and receive signals. This had 11 communication channels that are unlicensed. Newer wireless devices can also use the 5 GHZ band which has 24 unlicensed bands Narrowband, Broadband, and Spread Spectrum Signals Narrowband – a transmitter concentrates the signal energy at a single frequency or in a very small range of frequencies Broadband – uses a relatively wide band of the wireless spectrum and offers higher throughputs than narrowband technologies The use of multiple frequencies to transmit a signal is known as spread-spectrum technology. In other words a signal never stays continuously within one frequency range during its transmission. One specific implementation of spread spectrum is FHSS (frequency hoping spread spectrum). Another type is known as DSS (direct sequence spread spectrum) Fixed vs. Mobile Each type of wireless communication falls into one of two categories Fixed – the location of the transmitted and receiver do not move (results in energy saved because weaker signal strength is possible with directional antennas) Mobile – the location can change WLAN (Wireless LAN) Architecture There are two main types of arrangements Adhoc – data is sent directly between devices – good for small local devices Infrastructure mode – a wireless access point is placed centrally, that all devices connect with 802.11 WLANs The most popular wireless standards used on contemporary LANs are those developed by IEEE’s 802.11 committee. Over the years several distinct standards related to wireless networking have been released. Four of the best known standards are also referred to as Wi-Fi. They are…. 802.11b 802.11a 802.11g 802.11n These four standards share many characteristics. i.e. All 4 use half duplex signalling Follow the same access method Access Method 802.11 standards specify the use of CSMA/CA (Carrier Sense Multiple Access with Collision Avoidance) to access a shared medium. Using CSMA/CA before a station begins to send data on an 802.11 network, it checks for existing wireless transmissions. If the source node detects no transmission activity on the network, it waits a brief period of time and then sends its transmission. If the source does detect activity, it waits a brief period of time before checking again. The destination node receives the transmission and, after verifying its accuracy, issues an acknowledgement (ACT) packet to the source. If the source receives the ACK it assumes the transmission was successful, – if it does not receive an ACK it assumes the transmission failed and sends it again. Association Two types of scanning… Active – station transmits a special frame, known as a prove, on all available channels within its frequency range. When an access point finds the probe frame, it issues a probe response. Passive – wireless station listens on all channels within its frequency range for a special signal, known as a beacon frame, issued from an access point – the beacon frame contains information necessary to connect to the point. Re-association occurs when a mobile user moves out of one access point’s range and into the range of another. Frames Read page 378 – 381 about frames and specific 802.11 protocols Bluetooth Networks Sony Ericson originally invented the Bluetooth technology in the early 1990s. In 1998 other manufacturers joined Ericsson in the Special Interest Group (SIG) whose aim was to refine and standardize the technology. Bluetooth was designed to be used on small networks composed of personal communications devices. It has become popular wireless technology for communicating among cellular telephones, phone headsets, etc. Wireless WANs and Internet Access Refer to pages 396 – 402 of the textbook for details.

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• Data management in unexpected places

  - by Ashok_Ora

  Normal 0 false false false EN-US X-NONE X-NONE Data management in unexpected places When you think of network switches, routers, firewall appliances, etc., it may not be obvious that at the heart of these kinds of solutions is an engine that can manage huge amounts of data at very high throughput with low latencies and high availability. Consider a network router that is processing tens (or hundreds) of thousands of network packets per second. So what really happens inside a router? Packets are streaming in at the rate of tens of thousands per second. Each packet has multiple attributes, for example, a destination, associated SLAs etc. For each packet, the router has to determine the address of the next “hop” to the destination; it has to determine how to prioritize this packet. If it’s a high priority packet, then it has to be sent on its way before lower priority packets. As a consequence of prioritizing high priority packets, lower priority data packets may need to be temporarily stored (held back), but addressed fairly. If there are security or privacy requirements associated with the data packet, those have to be enforced. You probably need to keep track of statistics related to the packets processed (someone’s sure to ask). You have to do all this (and more) while preserving high availability i.e. if one of the processors in the router goes down, you have to have a way to continue processing without interruption (the customer won’t be happy with a “choppy” VoIP conversation, right?). And all this has to be achieved without ANY intervention from a human operator – the router is most likely to be in a remote location – it must JUST CONTINUE TO WORK CORRECTLY, even when bad things happen. How is this implemented? As soon as a packet arrives, it is interpreted by the receiving software. The software decodes the packet headers in order to determine the destination, kind of packet (e.g. voice vs. data), SLAs associated with the “owner” of the packet etc. It looks up the internal database of “rules” of how to process this packet and handles the packet accordingly. The software might choose to hold on to the packet safely for some period of time, if it’s a low priority packet. Ah – this sounds very much like a database problem. For each packet, you have to minimally · Look up the most efficient next “hop” towards the destination. The “most efficient” next hop can change, depending on latency, availability etc. · Look up the SLA and determine the priority of this packet (e.g. voice calls get priority over data ftp) · Look up security information associated with this data packet. It may be necessary to retrieve the context for this network packet since a network packet is a small “slice” of a session. The context for the “header” packet needs to be stored in the router, in order to make this work. · If the priority of the packet is low, then “store” the packet temporarily in the router until it is time to forward the packet to the next hop. · Update various statistics about the packet. In most cases, you have to do all this in the context of a single transaction. For example, you want to look up the forwarding address and perform the “send” in a single transaction so that the forwarding address doesn’t change while you’re sending the packet. So, how do you do all this? Berkeley DB is a proven, reliable, high performance, highly available embeddable database, designed for exactly these kinds of usage scenarios. Berkeley DB is a robust, reliable, proven solution that is currently being used in these scenarios. First and foremost, Berkeley DB (or BDB for short) is very very fast. It can process tens or hundreds of thousands of transactions per second. It can be used as a pure in-memory database, or as a disk-persistent database. BDB provides high availability – if one board in the router fails, the system can automatically failover to another board – no manual intervention required. BDB is self-administering – there’s no need for manual intervention in order to maintain a BDB application. No need to send a technician to a remote site in the middle of nowhere on a freezing winter day to perform maintenance operations. BDB is used in over 200 million deployments worldwide for the past two decades for mission-critical applications such as the one described here. You have a choice of spending valuable resources to implement similar functionality, or, you could simply embed BDB in your application and off you go! I know what I’d do – choose BDB, so I can focus on my business problem. What will you do? /* Style Definitions */ table.MsoNormalTable {mso-style-name:"Table Normal"; mso-tstyle-rowband-size:0; mso-tstyle-colband-size:0; mso-style-noshow:yes; mso-style-priority:99; mso-style-qformat:yes; mso-style-parent:""; mso-padding-alt:0in 5.4pt 0in 5.4pt; mso-para-margin-top:0in; mso-para-margin-right:0in; mso-para-margin-bottom:10.0pt; mso-para-margin-left:0in; line-height:115%; mso-pagination:widow-orphan; font-size:11.0pt; font-family:"Calibri","sans-serif"; mso-ascii-font-family:Calibri; mso-ascii-theme-font:minor-latin; mso-hansi-font-family:Calibri; mso-hansi-theme-font:minor-latin;}

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• Most Innovative IDM Projects: Awards at OpenWorld

  - by Tanu Sood

  On Tuesday at Oracle OpenWorld 2012, Oracle recognized the winners of Innovation Awards 2012 at a ceremony presided over by Hasan Rizvi, Executive Vice President at Oracle. Oracle Fusion Middleware Innovation Awards recognize customers for achieving significant business value through innovative uses of Oracle Fusion Middleware offerings. Winners are selected based on the uniqueness of their business case, business benefits, level of impact relative to the size of the organization, complexity and magnitude of implementation, and the originality of architecture. This year’s Award honors customers for their cutting-edge solutions driving business innovation and IT modernization using Oracle Fusion Middleware. The program has grown over the past 6 years, receiving a record number of nominations from customers around the globe. The winners were selected by a panel of judges that ranked each nomination across multiple different scoring categories. Congratulations to both Avea and ETS for winning this year’s Innovation Award for Identity Management. Identity Management Innovation Award 2012 Winner – Avea Company: Founded in 2004, AveA is the sole GSM 1800 mobile operator of Turkey and has reached a nationwide customer base of 12.8 million as of the end of 2011 Region: Turkey (EMEA) Products: Oracle Identity Manager, Oracle Identity Analytics, Oracle Access Management Suite Business Drivers: ·         To manage the agility and scale required for GSM Operations and enable call center efficiency by enabling agents to change their identity profiles (accounts and entitlements) rapidly based on call load. ·         Enhance user productivity and call center efficiency with self service password resets ·         Enforce compliance and audit reporting ·         Seamless identity management between AveA and parent company Turk Telecom Innovation and Results: ·         One of the first Sun2Oracle identity management migrations designed for high performance provisioning and trusted reconciliation built with connectors developed on the ICF architecture that provides custom user interfaces for  dynamic and rapid management of roles and entitlements along with entitlement level attestation using closed loop remediation between Oracle Identity Manager and Oracle Identity Analytics. ·         Dramatic reduction in identity administration and call center password reset tasks leading to 20% reduction in administration costs and 95% reduction in password related calls. ·         Enhanced user productivity by up to 25% to date ·         Enforced enterprise security and reduced risk ·         Cost-effective compliance management ·         Looking to seamlessly integrate with parent and sister companies’ infrastructure securely. Identity Management Innovation Award 2012 Winner – Education Testing Service (ETS)       See last year's winners here --Company: ETS is a private nonprofit organization devoted to educational measurement and research, primarily through testing. Region: U.S.A (North America) Products: Oracle Access Manager, Oracle Identity Federation, Oracle Identity Manager Business Drivers: ETS develops and administers more than 50 million achievement and admissions tests each year in more than 180 countries, at more than 9,000 locations worldwide.  As the business becomes more globally based, having a robust solution to security and user management issues becomes paramount. The organizations was looking for: ·         Simplified user experience for over 3000 company users and more than 6 million dynamic student and staff population ·         Infrastructure and administration cost reduction ·         Managing security risk by controlling 3rd party access to ETS systems ·         Enforce compliance and manage audit reporting ·         Automate on-boarding and decommissioning of user account to improve security, reduce administration costs and enhance user productivity ·         Improve user experience with simplified sign-on and user self service Innovation and Results: 1.    Manage Risk ·         Centralized system to control user access ·         Provided secure way of accessing service providers' application using federated SSO. ·         Provides reporting capability for auditing, governance and compliance. 2.    Improve efficiency ·         Real-Time provisioning to target systems ·         Centralized provisioning system for user management and access controls. ·         Enabling user self services. 3.    Reduce cost ·         Re-using common shared services for provisioning, SSO, Access by application reducing development cost and time. ·         Reducing infrastructure and maintenance cost by decommissioning legacy/redundant IDM services. ·         Reducing time and effort to implement security functionality in business applications (“onboard” instead of new development). ETS was able to fold in new and evolving requirement in addition to the initial stated goals realizing quick ROI and successfully meeting business objectives. Congratulations to the winners once again. We will be sure to bring you more from these Innovation Award winners over the next few months.

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• Cloud Computing Forces Better Design Practices

  - by Herve Roggero

  Is cloud computing simply different than on premise development, or is cloud computing actually forcing you to create better applications than you normally would? In other words, is cloud computing merely imposing different design principles, or forcing better design principles?  A little while back I got into a discussion with a developer in which I was arguing that cloud computing, and specifically Windows Azure in his case, was forcing developers to adopt better design principles. His opinion was that cloud computing was not yielding better systems; just different systems. In this blog, I will argue that cloud computing does force developers to use better design practices, and hence better applications. So the first thing to define, of course, is the word “better”, in the context of application development. Looking at a few definitions online, better means “superior quality”. As it relates to this discussion then, I stipulate that cloud computing can yield higher quality applications in terms of scalability, everything else being equal. Before going further I need to also outline the difference between performance and scalability. Performance and scalability are two related concepts, but they don’t mean the same thing. Scalability is the measure of system performance given various loads. So when developers design for performance, they usually give higher priority to a given load and tend to optimize for the given load. When developers design for scalability, the actual performance at a given load is not as important; the ability to ensure reasonable performance regardless of the load becomes the objective. This can lead to very different design choices. For example, if your objective is to obtains the fastest response time possible for a service you are building, you may choose the implement a TCP connection that never closes until the client chooses to close the connection (in other words, a tightly coupled service from a connectivity standpoint), and on which a connection session is established for faster processing on the next request (like SQL Server or other database systems for example). If you objective is to scale, you may implement a service that answers to requests without keeping session state, so that server resources are released as quickly as possible, like a REST service for example. This alternate design would likely have a slower response time than the TCP service for any given load, but would continue to function at very large loads because of its inherently loosely coupled design. An example of a REST service is the NO-SQL implementation in the Microsoft cloud called Azure Tables. Now, back to cloud computing… Cloud computing is designed to help you scale your applications, specifically when you use Platform as a Service (PaaS) offerings. However it’s not automatic. You can design a tightly-coupled TCP service as discussed above, and as you can imagine, it probably won’t scale even if you place the service in the cloud because it isn’t using a connection pattern that will allow it to scale [note: I am not implying that all TCP systems do not scale; I am just illustrating the scalability concepts with an imaginary TCP service that isn’t designed to scale for the purpose of this discussion]. The other service, using REST, will have a better chance to scale because, by design, it minimizes resource consumption for individual requests and doesn’t tie a client connection to a specific endpoint (which means you can easily deploy this service to hundreds of machines without much trouble, as long as your pockets are deep enough). The TCP and REST services discussed above are both valid designs; the TCP service is faster and the REST service scales better. So is it fair to say that one service is fundamentally better than the other? No; not unless you need to scale. And if you don’t need to scale, then you don’t need the cloud in the first place. However, it is interesting to note that if you do need to scale, then a loosely coupled system becomes a better design because it can almost always scale better than a tightly-coupled system. And because most applications grow overtime, with an increasing user base, new functional requirements, increased data and so forth, most applications eventually do need to scale. So in my humble opinion, I conclude that a loosely coupled system is not just different than a tightly coupled system; it is a better design, because it will stand the test of time. And in my book, if a system stands the test of time better than another, it is of superior quality. Because cloud computing demands loosely coupled systems so that its underlying service architecture can be leveraged, developers ultimately have no choice but to design loosely coupled systems for the cloud. And because loosely coupled systems are better… … the cloud forces better design practices. My 2 cents.

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• How can I implement a database TableView like thing in C++?

  - by Industrial-antidepressant

  How can I implement a TableView like thing in C++? I want to emulating a tiny relation database like thing in C++. I have data tables, and I want to transform it somehow, so I need a TableView like class. I want filtering, sorting, freely add and remove items and transforming (ex. view as UPPERCASE and so on). The whole thing is inside a GUI application, so datatables and views are attached to a GUI (or HTML or something). So how can I identify an item in the view? How can I signal it when the table is changed? Is there some design pattern for this? Here is a simple table, and a simple data item: #include <string> #include <boost/multi_index_container.hpp> #include <boost/multi_index/member.hpp> #include <boost/multi_index/ordered_index.hpp> #include <boost/multi_index/random_access_index.hpp> using boost::multi_index_container; using namespace boost::multi_index; struct Data { Data() {} int id; std::string name; }; struct row{}; struct id{}; struct name{}; typedef boost::multi_index_container< Data, indexed_by< random_access<tag<row> >, ordered_unique<tag<id>, member<Data, int, &Data::id> >, ordered_unique<tag<name>, member<Data, std::string, &Data::name> > > > TDataTable; class DataTable { public: typedef Data item_type; typedef TDataTable::value_type value_type; typedef TDataTable::const_reference const_reference; typedef TDataTable::index<row>::type TRowIndex; typedef TDataTable::index<id>::type TIdIndex; typedef TDataTable::index<name>::type TNameIndex; typedef TRowIndex::iterator iterator; DataTable() : row_index(rule_table.get<row>()), id_index(rule_table.get<id>()), name_index(rule_table.get<name>()), row_index_writeable(rule_table.get<row>()) { } TDataTable::const_reference operator[](TDataTable::size_type n) const { return rule_table[n]; } std::pair<iterator,bool> push_back(const value_type& x) { return row_index_writeable.push_back(x); } iterator erase(iterator position) { return row_index_writeable.erase(position); } bool replace(iterator position,const value_type& x) { return row_index_writeable.replace(position, x); } template<typename InputIterator> void rearrange(InputIterator first) { return row_index_writeable.rearrange(first); } void print_table() const; unsigned size() const { return row_index.size(); } TDataTable rule_table; const TRowIndex& row_index; const TIdIndex& id_index; const TNameIndex& name_index; private: TRowIndex& row_index_writeable; }; class DataTableView { DataTableView(const DataTable& source_table) {} // How can I implement this? // I want filtering, sorting, signaling upper GUI layer, and sorting, and ... }; int main() { Data data1; data1.id = 1; data1.name = "name1"; Data data2; data2.id = 2; data2.name = "name2"; DataTable table; table.push_back(data1); DataTable::iterator it1 = table.row_index.iterator_to(table[0]); table.erase(it1); table.push_back(data1); Data new_data(table[0]); new_data.name = "new_name"; table.replace(table.row_index.iterator_to(table[0]), new_data); for (unsigned i = 0; i < table.size(); ++i) std::cout << table[i].name << std::endl; #if 0 // using scenarios: DataTableView table_view(table); table_view.fill_from_source(); // synchronization with source table_view.remove(data_item1); // remove item from view table_view.add(data_item2); // add item from source table table_view.filter(filterfunc); // filtering table_view.sort(sortfunc); // sorting // modifying from source_able, hot to signal the table_view? // FYI: Table view is atteched to a GUI item table.erase(data); table.replace(data); #endif return 0; }

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• Deduping your redundancies

  - by nospam(at)example.com (Joerg Moellenkamp)

  Robin Harris of Storagemojo pointed to an interesting article about about deduplication and it's impact to the resiliency of your data against data corruption on ACM Queue. The problem in short: A considerable number of filesystems store important metadata at multiple locations. For example the ZFS rootblock is copied to three locations. Other filesystems have similar provisions to protect their metadata. However you can easily proof, that the rootblock pointer in the uberblock of ZFS for example is pointing to blocks with absolutely equal content in all three locatition (with zdb -uu and zdb -r). It has to be that way, because they are protected by the same checksum. A number of devices offer block level dedup, either as an option or as part of their inner workings. However when you store three identical blocks on them and the devices does block level dedup internally, the device may just deduplicated your redundant metadata to a block stored just once that is stored on the non-voilatile storage. When this block is corrupted, you have essentially three corrupted copies. Three hit with one bullet. This is indeed an interesting problem: A device doing deduplication doesn't know if a block is important or just a datablock. This is the reason why I like deduplication like it's done in ZFS. It's an integrated part and so important parts don't get deduplicated away. A disk accessed by a block level interface doesn't know anything about the importance of a block. A metadata block is nothing different to it's inner mechanism than a normal data block because there is no way to tell that this is important and that those redundancies aren't allowed to fall prey to some clever deduplication mechanism. Robin talks about this in regard of the Sandforce disk controllers who use a kind of dedup to reduce some of the nasty effects of writing data to flash, but the problem is much broader. However this is relevant whenever you are using a device with block level deduplication. It's just the point that you have to activate it for most implementation by command, whereas certain devices do this by default or by design and you don't know about it. However I'm not perfectly sure about that ? given that storage administration and server administration are often different groups with different business objectives I would ask your storage guys if they have activated dedup without telling somebody elase on their boxes in order to speak less often with the storage sales rep. The problem is even more interesting with ZFS. You may use ditto blocks to protect important data to store multiple copies of data in the pool to increase redundancy, even when your pool just consists out of one disk or just a striped set of disk. However when your device is doing dedup internally it may remove your redundancy before it hits the nonvolatile storage. You've won nothing. Just spend your disk quota on the the LUNs in the SAN and you make your disk admin happy because of the good dedup ratio However you can just fall in this specific "deduped ditto block"trap when your pool just consists out of a single device, because ZFS writes ditto blocks on different disks, when there is more than just one disk. Yet another reason why you should spend some extra-thought when putting your zpool on a single LUN, especially when the LUN is sliced and dices out of a large heap of storage devices by a storage controller. However I have one problem with the articles and their specific mention of ZFS: You can just hit by this problem when you are using the deduplicating device for the pool. However in the specifically mentioned case of SSD this isn't the usecase. Most implementations of SSD in conjunction with ZFS are hybrid storage pools and so rotating rust disk is used as pool and SSD are used as L2ARC/sZIL. And there it simply doesn't matter: When you really have to resort to the sZIL (your system went down, it doesn't matter of one block or several blocks are corrupt, you have to fail back to the last known good transaction group the device. On the other side, when a block in L2ARC is corrupt, you simply read it from the pool and in HSP implementations this is the already mentioned rust. In conjunction with ZFS this is more interesting when using a storage array, that is capable to do dedup and where you use LUNs for your pool. However as mentioned before, on those devices it's a user made decision to do so, and so it's less probable that you deduplicating your redundancies. Other filesystems lacking acapability similar to hybrid storage pools are more "haunted" by this problem of SSD using dedup-like mechanisms internally, because those filesystem really store the data on the the SSD instead of using it just as accelerating devices. However at the end Robin is correct: It's jet another point why protecting your data by creating redundancies by dispersing it several disks (by mirror or parity RAIDs) is really important. No dedup mechanism inside a device can dedup away your redundancy when you write it to a totally different and indepenent device.

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• Determining the angle to fire a shot when target and shooter moves, and bullet moves with shooter velocity added in

  - by Azaral

  I saw this question: Predicting enemy position in order to have an object lead its target and followed the link in the answer to stack overflow. In the stack overflow page I used the 2nd answer, the one that is a large mathematical derivation. My situation is a little different though. My first question though is will the answer provided in the stack overflow page even work to begin with, assuming the original circumstances of moving target and stationary shooter. My situation is a little different than that situation. My target moves, the shooter moves, and the bullets from the shooter start off with the velocities in x and y added to the bullets' x and y velocities. If you are sliding to the right, the bullets will remain in front of you as you move so as long as your velocity remains constant. What I'm trying to do is to get the enemy to be able to determine where they need to shoot in order to hit the player. Unless the player and enemy is stationary, the velocity from the ship adding to the velocity of the bullets will cause a miss. I'd rather like to prevent that. I used the formula in the stack overflow answer and did what I thought were the appropriate adjustments. I've been banging at this for the last four hours and I just can't make it click. It is probably something really simple and boneheaded that I am missing (that seems to be a lot of my problems lately). Here is the solution presented from the stack overflow answer: It boils down to solving a quadratic equation of the form: a * sqr(x) + b * x + c == 0 Note that by sqr I mean square, as opposed to square root. Use the following values: a := sqr(target.velocityX) + sqr(target.velocityY) - sqr(projectile_speed) b := 2 * (target.velocityX * (target.startX - cannon.X) + target.velocityY * (target.startY - cannon.Y)) c := sqr(target.startX - cannon.X) + sqr(target.startY - cannon.Y) Now we can look at the discriminant to determine if we have a possible solution. disc := sqr(b) - 4 * a * c If the discriminant is less than 0, forget about hitting your target -- your projectile can never get there in time. Otherwise, look at two candidate solutions: t1 := (-b + sqrt(disc)) / (2 * a) t2 := (-b - sqrt(disc)) / (2 * a) Note that if disc == 0 then t1 and t2 are equal. If there are no other considerations such as intervening obstacles, simply choose the smaller positive value. (Negative t values would require firing backward in time to use!) Substitute the chosen t value back into the target's position equations to get the coordinates of the leading point you should be aiming at: aim.X := t * target.velocityX + target.startX aim.Y := t * target.velocityY + target.startY Here is my code, after being corrected by Sam Hocevar (thank you again for your help!). It still doesn't work. For some reason it never enters the section of code inside the if(disc = 0) (obviously because it is always less than zero but...). However, if I plug the numbers from my game log on the enemy and player positions and velocities it outputs a valid firing solution. I have looked at the code side by side a couple of times now and I can't find any differences. There has got to be something simple I'm missing here. If someone else could look at this code and determine what is going on here I'd appreciate it. I know it's not going through that section because if it were, shouldShoot would become true and the enemy would be blasting away at the player. This section calls the function in question, CalculateShootHeading() if(shouldMove) { UseEngines(); } x += xVelocity; y += yVelocity; CalculateShootHeading(); if(shouldShoot) { ShootWeapons(); } UpdateWeapons(); This is CalculateShootHeading(). This is inside the enemy class so x and y are the enemy's x and y and the same with velocity. One output from my game log gives Player X = 2108, Player Y = -180.956, Player X velocity = 10.9949, Player Y Velocity = -6.26017, Enemy X = 1988.31, Enemy Y = -339.051, Enemy X velocity = 1.81666, Enemy Y velocity = -9.67762, 0 enemy projectiles. The output from the console tester is Bullet position = 2210.49, -239.313 and Player Position = 2210.49, -239.313. This doesn't make any sense. The only thing that could be different is the code or the input into my function in the game and I've checked that and I don't think that it is wrong as it's updated before this and never changed. float const bulletSpeed = 30.f; float const dx = playerX - x; float const dy = playerY - y; float const vx = playerXVelocity - xVelocity; float const vy = playerYVelocity - yVelocity; float const a = vx * vx + vy * vy - bulletSpeed * bulletSpeed; float const b = 2.f * (vx * dx + vy * dy); float const c = dx * dx + dy * dy; float const disc = b * b - 4.f * a * c; shouldShoot = false; if (disc >= 0.f) { float t0 = (-b - std::sqrt(disc)) / (2.f * a); float t1 = (-b + std::sqrt(disc)) / (2.f * a); if (t0 < 0.f || (t1 < t0 && t1 >= 0.f)) { t0 = t1; } if (t0 >= 0.f) { float shootx = vx + dx / t0; float shooty = vy + dy / t0; heading = std::atan2(shooty, shootx) * RAD2DEGREE; } shouldShoot = true; }

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• C++ strongly typed typedef

  - by Kian

  I've been trying to think of a way of declaring strongly typed typedefs, to catch a certain class of bugs in the compilation stage. It's often the case that I'll typedef an int into several types of ids, or a vector to position or velocity: typedef int EntityID; typedef int ModelID; typedef Vector3 Position; typedef Vector3 Velocity; This can make the intent of code more clear, but after a long night of coding one might make silly mistakes like comparing different kinds of ids, or adding a position to a velocity perhaps. EntityID eID; ModelID mID; if ( eID == mID ) // <- Compiler sees nothing wrong { /*bug*/ } Position p; Velocity v; Position newP = p + v; // bug, meant p + v*s but compiler sees nothing wrong Unfortunately, suggestions I've found for strongly typed typedefs include using boost, which at least for me isn't a possibility (I do have c++11 at least). So after a bit of thinking, I came upon this idea, and wanted to run it by someone. First, you declare the base type as a template. The template parameter isn't used for anything in the definition, however: template < typename T > class IDType { unsigned int m_id; public: IDType( unsigned int const& i_id ): m_id {i_id} {}; friend bool operator==<T>( IDType<T> const& i_lhs, IDType<T> const& i_rhs ); }; Friend functions actually need to be forward declared before the class definition, which requires a forward declaration of the template class. We then define all the members for the base type, just remembering that it's a template class. Finally, when we want to use it, we typedef it as: class EntityT; typedef IDType<EntityT> EntityID; class ModelT; typedef IDType<ModelT> ModelID; The types are now entirely separate. Functions that take an EntityID will throw a compiler error if you try to feed them a ModelID instead, for example. Aside from having to declare the base types as templates, with the issues that entails, it's also fairly compact. I was hoping anyone had comments or critiques about this idea? One issue that came to mind while writing this, in the case of positions and velocities for example, would be that I can't convert between types as freely as before. Where before multiplying a vector by a scalar would give another vector, so I could do: typedef float Time; typedef Vector3 Position; typedef Vector3 Velocity; Time t = 1.0f; Position p = { 0.0f }; Velocity v = { 1.0f, 0.0f, 0.0f }; Position newP = p + v*t; With my strongly typed typedef I'd have to tell the compiler that multypling a Velocity by a Time results in a Position. class TimeT; typedef Float<TimeT> Time; class PositionT; typedef Vector3<PositionT> Position; class VelocityT; typedef Vector3<VelocityT> Velocity; Time t = 1.0f; Position p = { 0.0f }; Velocity v = { 1.0f, 0.0f, 0.0f }; Position newP = p + v*t; // Compiler error To solve this, I think I'd have to specialize every conversion explicitly, which can be kind of a bother. On the other hand, this limitation can help prevent other kinds of errors (say, multiplying a Velocity by a Distance, perhaps, which wouldn't make sense in this domain). So I'm torn, and wondering if people have any opinions on my original issue, or my approach to solving it.

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• [C#] Convert string to double with 2 digit after decimal separator

  - by st.stoqnov

  All began with these simple lines of code: string s = "16.9"; double d = Convert.ToDouble(s); d*=100; The result should be 1690.0, but it's not. d is equal to 1689.9999999999998. All I want to do is to round a double to value with 2 digit after decimal separator. Here is my function. private double RoundFloat(double Value) { float sign = (Value < 0) ? -0.01f : 0.01f; if (Math.Abs(Value) < 0.00001) Value = 0; string SVal = Value.ToString(); string DecimalSeparator = System.Globalization.CultureInfo.CurrentCulture.NumberFormat.CurrencyDecimalSeparator; int i = SVal.IndexOf(DecimalSeparator); if (i > 0) { int SRnd; try { // ????? ??????? ????? ???? ?????????? ?????????? SRnd = Convert.ToInt32(SVal.Substring(i + 3, 1)); } catch { SRnd = 0; } if (SVal.Length > i + 3) SVal = SVal.Substring(0, i + 3); //SVal += "00001"; try { double result = (SRnd >= 5) ? Convert.ToDouble(SVal) + sign : Convert.ToDouble(SVal); //result = Math.Round(result, 2); return result; } catch { return 0; } } else { return Value; } But again the same problem, converting from string to double is not working as I want. A workaround to this problem is to concatenate "00001" to the string and then use the Math.Round function (commented in the example above). This double value multiplied to 100 (as integer) is send to a device (cash register) and this values must be correct. I am using VS2005 + .NET CF 2.0 Is there another more "elegant" solution, I am not happy with this one.

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• Runge-Kutta (RK4) integration for game physics

  - by Kai

  Gaffer on Games has a great article about using RK4 integration for better game physics. The implementation is straightforward but the math behind it confuses me. I understand derivatives and integrals on a conceptual level but I haven't manipulated equations in a long time. Here's the brunt of Gaffer's implementation: void integrate(State &state, float t, float dt) { Derivative a = evaluate(state, t, 0.0f, Derivative()); Derivative b = evaluate(state, t+dt*0.5f, dt*0.5f, a); Derivative c = evaluate(state, t+dt*0.5f, dt*0.5f, b); Derivative d = evaluate(state, t+dt, dt, c); const float dxdt = 1.0f/6.0f * (a.dx + 2.0f*(b.dx + c.dx) + d.dx); const float dvdt = 1.0f/6.0f * (a.dv + 2.0f*(b.dv + c.dv) + d.dv) state.x = state.x + dxdt * dt; state.v = state.v + dvdt * dt; } Can anybody explain in simple terms how RK4 works? Specifically, why are we averaging the derivatives at 0.0f, 0.5f, 0.5f, and 1.0f? How is averaging derivatives up to the 4th order different from doing a simple euler integration with a smaller timestep? After reading the accepted answer below, and several other articles, I have a grasp on how RK4 works. To answer my own questions: Can anybody explain in simple terms how RK4 works? RK4 takes advantage of the fact that we can get a much better approximation of a function if we use its higher-order derivatives rather than just the first or second derivative. That's why the Taylor series converges much faster than Euler approximations. (take a look at the animation on the right side of that page) Specifically, why are we averaging the derivatives at 0.0f, 0.5f, 0.5f, and 1.0f? The Runge-Kutta method is an approximation of a function that samples derivatives of several points within a timestep, unlike the Taylor series which only samples derivatives of a single point. After sampling these derivatives we need to know how to weigh each sample to get the closest approximation possible. An easy way to do this is to pick constants that coincide with the Taylor series, which is how the constants of a Runge-Kutta equation are determined. This article made it clearer for me: http://web.mit.edu/10.001/Web/Course%5FNotes/Differential%5FEquations%5FNotes/node5.html. Notice how (15) is the Taylor series expansion while (17) is the Runge-Kutta derivation. How is averaging derivatives up to the 4th order different from doing a simple euler integration with a smaller timestep? Mathematically it converges much faster than doing many Euler approximations. Of course, with enough Euler approximations we can gain equal accuracy to RK4, but the computational power needed doesn't justify using Euler.

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• InteropServices COMException when executing a .net app from a web CGI script on Windows Server 2003

  - by Kurt W. Leucht

  Disclaimer: I'm completely clueless about .net and COM. I have a vendor's application that appears to be written in .net and I'm trying to wrap it with a web form (a cgi-bin Perl script) so I can eventually launch this vendor's app from a separate computer. I'm on a Windows Server 2003 R2 SE SP1 system and I'm using Apache 2.2 for the web server and ActivePerl 5.10.0.1004 for the cgi script. My cgi script calls the vendor's app that resides on the same machine using the Perl backtick operator. ... $result = "Result: " . `$vendorsPath/$vendorsExecutable $arg1 $arg2`; ... Right now I'm just running IE web browser locally on the server machine and accessing "http://localhost/cgi-bin/myPerlScript.pl". The vendor's app fails and logs a debug message that includes the following stack trace (I changed a couple names so as to not give away the vendor's identity): ... System.Reflection.TargetInvocationException: Exception has been thrown by the target of an invocation. ---> System.Runtime.InteropServices.COMException (0x80043A1D): 0x80040154 - Class not registered --- End of inner exception stack trace --- at System.RuntimeType.InvokeDispMethod(String name, BindingFlags invokeAttr, Object target, Object[] args, Boolean[] byrefModifiers, Int32 culture, String[] namedParameters) at System.RuntimeType.InvokeMember(String name, BindingFlags invokeAttr, Binder binder, Object target, Object[] args, ParameterModifier[] modifiers, CultureInfo culture, String[] namedParameters) at VendorsTool.Engine.Core.VendorsEngine.LoadVendorsServices(String fileName, String& projectCommPath) ... When I run the vendors app from the Windows command line on the server machine with the exact same arguments that the cgi script is passing it runs just fine, so there's something about invoking their app via the web script that is causing a problem. This problem is likely security related because the whole thing runs just fine on a Windows XP Pro machine (both command line and web invocation). I actually developed my web script there and got it completely working there before I tried moving it to the Windows Server 2003 machine. So what's different about the Windows Server 2003 machine that would keep the vendor's .net app from being executed successfully by a web cgi script? Can I fix this problem somehow to make it work on my server or will the vendor have to make a change to their .net app and ship out a new version? I'm probably the only person in the world who is trying to execute this vendor's app from a separate program, so I hate to bother the vendor with the issue if there's a workaround that I can implement myself here on my server machine. Plus, I'm in kind of a hurry and I don't want to wait 4 or 6 months for the vendor to put in a fix and deploy a new version. Thanks for any advise you can give.

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• Extracting pair member in lambda expressions and template typedef idiom

  - by Nazgob

  Hi, I have some complex types here so I decided to use nifty trick to have typedef on templated types. Then I have a class some_container that has a container as a member. Container is a vector of pairs composed of element and vector. I want to write std::find_if algorithm with lambda expression to find element that have certain value. To get the value I have to call first on pair and then get value from element. Below my std::find_if there is normal loop that does the trick. My lambda fails to compile. How to access value inside element which is inside pair? I use g++ 4.4+ and VS 2010 and I want to stick to boost lambda for now. #include <vector> #include <algorithm> #include <boost\lambda\lambda.hpp> #include <boost\lambda\bind.hpp> template<typename T> class element { public: T value; }; template<typename T> class element_vector_pair // idiom to have templated typedef { public: typedef std::pair<element<T>, std::vector<T> > type; }; template<typename T> class vector_containter // idiom to have templated typedef { public: typedef std::vector<typename element_vector_pair<T>::type > type; }; template<typename T> bool operator==(const typename element_vector_pair<T>::type & lhs, const typename element_vector_pair<T>::type & rhs) { return lhs.first.value == rhs.first.value; } template<typename T> class some_container { public: element<T> get_element(const T& value) const { std::find_if(container.begin(), container.end(), bind(&typename vector_containter<T>::type::value_type::first::value, boost::lambda::_1) == value); /*for(size_t i = 0; i < container.size(); ++i) { if(container.at(i).first.value == value) { return container.at(i); } }*/ return element<T>(); //whatever } protected: typename vector_containter<T>::type container; }; int main() { some_container<int> s; s.get_element(5); return 0; }

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• gcc 4.5 installation problem under ubuntu

  - by Claire Huang

  I tried to install gcc 4.5 on ubuntu 10.04 but failed. Here is a compile error that I don't know how to solve. Is there anyone successfully install the latest gcc on ubuntu? Following is my steps and the error message, I'd like to know where is the problem.... Step1: download these files: gcc-core-4.5.0.tar.gz gcc-g++-4.5.0.tar.gz gmp-4.3.2.tar.bz2 mpc-0.8.1.tar.gz mpfr-2.4.2.tar.gz Step2: Unzip above files Step3: move gmp, mpc, mpfr to the gcc-4.5.0/ directory. mv gmp-4.3.2 gcc-4.5.0/gmp mv mpc-0.8.1 gcc-4.5.0/mpc mv mpfr-2.4.2 gcc-4.5.0/mpfr Step4: go to gcc-4.5.0 directory and do configuration: sudo ./configure Step5: compile and install sudo make sudo make install The first 4 steps is OK, I can configure it successfully. However, when I try to compile it, following error message comes out, I cannot figure out what the problem is. Should I change the name from "gcc 4.5" to "gcc"?? It's a little strange that we need to do this by ourself. Is there anything I missed during the installation? xxx@xxx-laptop:/media/Data/Tool/linux/gcc 4.5/gcc-4.5.0$ sudo make [sudo] password for xxx: [ -f stage_final ] || echo stage3 > stage_final /bin/bash: line 2: test: /media/Data/Tool/linux/gcc: binary operator expected /bin/bash: /media/Data/Tool/linux/gcc: No such file or directory make[1]: Entering directory `/media/Data/Tool/linux/gcc 4.5/gcc-4.5.0' make[2]: Entering directory `/media/Data/Tool/linux/gcc 4.5/gcc-4.5.0' make[3]: Entering directory `/media/Data/Tool/linux/gcc 4.5/gcc-4.5.0' rm -f stage_current make[3]: Leaving directory `/media/Data/Tool/linux/gcc 4.5/gcc-4.5.0' make[2]: Leaving directory `/media/Data/Tool/linux/gcc 4.5/gcc-4.5.0' make[2]: Entering directory `/media/Data/Tool/linux/gcc 4.5/gcc-4.5.0' Configuring stage 1 in host-x86_64-unknown-linux-gnu/intl /bin/bash: /media/Data/Tool/linux/gcc: No such file or directory make[2]: *** [configure-stage1-intl] Error 127 make[2]: Leaving directory `/media/Data/Tool/linux/gcc 4.5/gcc-4.5.0' make[1]: *** [stage1-bubble] Error 2 make[1]: Leaving directory `/media/Data/Tool/linux/gcc 4.5/gcc-4.5.0' make: *** [all] Error 2

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• Comparing strings with user-created string class

  - by meepz

  Basically, I created my own string class, mystring.h and mystring.c. What I want to do is write a function that compares two "strings" and then returns 1 first word is larger than the second, the opposite if word 2 is larger than word 1, and 0 if the two words are equal. What I have so far is this: int compareto(void * S1, void * S2){ String s1 = (String S1); String s2 = (String S2); int i, cs1 = 0, cs2 = 0; //cs1 is count of s1, cs2 is count of s2 while(s1->c[i] != '\0'){ //basically, while there is a word if(s1->c[i] < s2->c[i]) // if string 1 char is less than string 2 char cs2++; //add to string 2 count else (s1->c[i] > s2->c[i]) //vice versa cs1++; i++; } for my return I basically have if(cs1>cs2){ return 1; } else if(cs2 > cs1){ return 2; } return 0; here is mystring.h typedef struct mystring { char * c; int length; int (*sLength)(void * s); char (*charAt)(void * s, int i); int (*compareTo)(void * s1, void * s2); struct mystring * (*concat)(void * s1, void * s2); struct mystring * (*subString)(void * s, int begin, int end); void (*printS)(void * s); } string_t; typedef string_t * String; This does what I want, but only for unspecified order. What I want to do is search through my linked list for the last name. Ex. I have two entries in my linked list, smith and jones; Jones will be output as greater than smith, but alphabetically it isnt. (I'm using this to remove student entries from a generic link list I created) Any suggestions, all of my google searches involve using the library, so I've had no luck)

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• Delete Nodes + attributes that match Xpath except specific attributes

  - by Ryan Ternier

  I'm trying to find the best (efficient) way of doing this. I have a medium sized XML document. Depending on specific settings certain portions of it need to be filtered out for security reasons. I'll be doing this in XSLT as it's configurable and no code should need changing. I've looked around, but not getting much luck on it. For example: I have the following XPath: //*[@root='2.16.840.1.113883.3.51.1.1.6.1'] Whicrooth gives me all nodes with a root attribute equal to a specific OID. In these nodes I want to have all attributes except for a few (ex. foo and bar) erased, and then having another attribute added (ex. reason) I also need to have multiple XPath expressions that can be ran to zero down on a specific node and clear it's contents out in a similar fashion, with respect to nodes with specific attributes. I'm playing around with information from: XPath expression to select all XML child nodes except a specific list? and XSLT Remove Elements and/or Attributes by Name per XSL Parameters Will update shortly when I can have access what what I"ve done so far. Example: XML Before Transformation <root> <childNode> <innerChild root="2.16.840.1.113883.3.51.1.1.6.1" a="b" b="c" type="innerChildness"/> <innerChildSibling/> </childNode> <animals> <cat> <name>bob</name> </cat> </animals> <tree/> <water root="2.16.840.1.113883.3.51.1.1.6.1" z="zed" l="ell" type="liquidLIke"/> </root> After <root> <childNode> <innerChild root="2.16.840.1.113883.3.51.1.1.6.1" flavor="MSK"/> <!-- filtered --> <innerChildSibling/> </childNode> <animals> <cat flavor="MSK" /> <!-- cat was filtered --> </animals> <tree/> <water root="2.16.840.1.113883.3.51.1.1.6.1" flavor="MSK"/> <!-- filtered --> </root>

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• SIFR 3.0 - Font Size

  - by Nick

  I have been working with SIFR 3.0 for some time now and the font-size never seems to work correctly. I understand the most basic concepts behind SIFR. SIFR runs when you load the page. It does some calculations one the size of the HTML rendered font and then replaces it with a flash movie that is roughly equal to that size. Because of this, you want to style your HTML font to match the size of your SIFR font as close as possible. My problem always comes up when trying to style these two font sizes to match. Let's say I want to use a SIFR font of Helvetica Neue Lt at about 32px. The HTML equivalent is something like Arial Narrow at about 36px with some negative letter spacing. So here is what I do. In sifr.css I'll write: @media screen { .sIFR-active h1 { visibility: hidden; z-index: 0 !important; font-size: 36px; } } Great, that gets the default HTML font the size I need. Now I need to update the flash SIFR font size. So I go into sifr-config.js and write something like this: sIFR.replace(HelveticaNeueThinCond, { selector: 'h1', css: '.sIFR-root { color: #762123; font-size: 32px; line-height: 1em; }', transparent: true }); So right now everything is working great. That is until my h1 text wraps more than one line. For some reason, when the text wraps it only shows the first line. It seems calculates the height wrong. This is very weird because I ran some tests. I took "visibility: hidden" off of "sIFR-active h1" to make sure that the HTML rendered text was the right size. It is, it takes up two lines. However, when the Flash replaces this text it gives it a min-height of one line of text. Odd. The only way I could find to fix this wrapping problem was to remove "font-size: 32px;" from "sIFR.replace(HelveticaNeueThinCond" in sifr-config. The problem I run into then is that it inherits the font-size set in sifr.css. Now the problem is that my HTML text is bigger then the SIFR text. So occasional my HTML text will wrap to a new line before my SIFR text leaving a big white space. So, how do I set two different font-size (one for my HTML text and one for my SIFR) without losing the wrapping. The only time I have been able to use the successfully is when I have a SIFR font that is so similar to a web safe font that they can share the same font-size attribute. Thanks

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• Hive NR map progress inconsistent and regurlarly restart from 0%

  - by user92471

  I have a Yarn MR (with two ec2 instances to mapreduce) job on a dataset of approximately a thousand avro records, and the map phase is behaving erratically. See the progress below. Of course i checked the logs on resourcemanager and nodemanagers and saw nothing suspicious, but these logs are too verbose What is going on there ? hive> select * from nikon where qs_cs_s_aid='VIEW' limit 10; Total MapReduce jobs = 1 Launching Job 1 out of 1 Number of reduce tasks is set to 0 since there's no reduce operator Starting Job = job_1352281315350_0020, Tracking URL = http://blabla.ec2.internal:8088/proxy/application_1352281315350_0020/ Kill Command = /usr/lib/hadoop/bin/hadoop job -Dmapred.job.tracker=blabla.com:8032 -kill job_1352281315350_0020 Hadoop job information for Stage-1: number of mappers: 4; number of reducers: 0 2012-11-07 11:14:40,976 Stage-1 map = 0%, reduce = 0% 2012-11-07 11:15:06,136 Stage-1 map = 1%, reduce = 0%, Cumulative CPU 10.38 sec 2012-11-07 11:15:07,253 Stage-1 map = 1%, reduce = 0%, Cumulative CPU 12.18 sec 2012-11-07 11:15:08,371 Stage-1 map = 1%, reduce = 0%, Cumulative CPU 12.18 sec 2012-11-07 11:15:09,491 Stage-1 map = 1%, reduce = 0%, Cumulative CPU 12.18 sec 2012-11-07 11:15:10,643 Stage-1 map = 2%, reduce = 0%, Cumulative CPU 15.42 sec (...) 2012-11-07 11:15:35,441 Stage-1 map = 28%, reduce = 0%, Cumulative CPU 37.77 sec 2012-11-07 11:15:36,486 Stage-1 map = 28%, reduce = 0%, Cumulative CPU 37.77 sec here restart at 16% ? 2012-11-07 11:15:37,692 Stage-1 map = 16%, reduce = 0%, Cumulative CPU 21.15 sec 2012-11-07 11:15:38,815 Stage-1 map = 16%, reduce = 0%, Cumulative CPU 21.15 sec 2012-11-07 11:15:39,865 Stage-1 map = 16%, reduce = 0%, Cumulative CPU 21.15 sec 2012-11-07 11:15:41,064 Stage-1 map = 18%, reduce = 0%, Cumulative CPU 22.4 sec 2012-11-07 11:15:42,181 Stage-1 map = 18%, reduce = 0%, Cumulative CPU 22.4 sec 2012-11-07 11:15:43,299 Stage-1 map = 18%, reduce = 0%, Cumulative CPU 22.4 sec here restart at 0% ? 2012-11-07 11:15:44,418 Stage-1 map = 0%, reduce = 0% 2012-11-07 11:16:02,076 Stage-1 map = 1%, reduce = 0%, Cumulative CPU 6.86 sec 2012-11-07 11:16:03,193 Stage-1 map = 1%, reduce = 0%, Cumulative CPU 6.86 sec 2012-11-07 11:16:04,259 Stage-1 map = 2%, reduce = 0%, Cumulative CPU 8.45 sec (...) 2012-11-07 11:16:31,291 Stage-1 map = 22%, reduce = 0%, Cumulative CPU 35.34 sec 2012-11-07 11:16:32,414 Stage-1 map = 26%, reduce = 0%, Cumulative CPU 37.93 sec here restart at 11% ? 2012-11-07 11:16:33,459 Stage-1 map = 11%, reduce = 0%, Cumulative CPU 19.53 sec 2012-11-07 11:16:34,507 Stage-1 map = 11%, reduce = 0%, Cumulative CPU 19.53 sec 2012-11-07 11:16:35,731 Stage-1 map = 13%, reduce = 0%, Cumulative CPU 21.47 sec (...) 2012-11-07 11:16:46,839 Stage-1 map = 17%, reduce = 0%, Cumulative CPU 24.14 sec here restart at 0% ? 2012-11-07 11:16:47,939 Stage-1 map = 0%, reduce = 0% 2012-11-07 11:16:56,653 Stage-1 map = 1%, reduce = 0%, Cumulative CPU 7.54 sec 2012-11-07 11:16:57,814 Stage-1 map = 1%, reduce = 0%, Cumulative CPU 7.54 sec (...) Needless to say the job crashes after some time with an Error: java.io.IOException: java.io.IOException: java.lang.ArrayIndexOutOfBoundsException: -56

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• Any significant performance improvement by using bitwise operators instead of plain int sums in C#?

  - by tunnuz

  Hello, I started working with C# a few weeks ago and I'm now in a situation where I need to build up a "bit set" flag to handle different cases in an algorithm. I have thus two options: enum RelativePositioning { LEFT = 0, RIGHT = 1, BOTTOM = 2, TOP = 3, FRONT = 4, BACK = 5 } pos = ((eye.X < minCorner.X ? 1 : 0) << RelativePositioning.LEFT) + ((eye.X > maxCorner.X ? 1 : 0) << RelativePositioning.RIGHT) + ((eye.Y < minCorner.Y ? 1 : 0) << RelativePositioning.BOTTOM) + ((eye.Y > maxCorner.Y ? 1 : 0) << RelativePositioning.TOP) + ((eye.Z < minCorner.Z ? 1 : 0) << RelativePositioning.FRONT) + ((eye.Z > maxCorner.Z ? 1 : 0) << RelativePositioning.BACK); Or: enum RelativePositioning { LEFT = 1, RIGHT = 2, BOTTOM = 4, TOP = 8, FRONT = 16, BACK = 32 } if (eye.X < minCorner.X) { pos += RelativePositioning.LEFT; } if (eye.X > maxCorner.X) { pos += RelativePositioning.RIGHT; } if (eye.Y < minCorner.Y) { pos += RelativePositioning.BOTTOM; } if (eye.Y > maxCorner.Y) { pos += RelativePositioning.TOP; } if (eye.Z > maxCorner.Z) { pos += RelativePositioning.FRONT; } if (eye.Z < minCorner.Z) { pos += RelativePositioning.BACK; } I could have used something as ((eye.X > maxCorner.X) << 1) but C# does not allow implicit casting from bool to int and the ternary operator was similar enough. My question now is: is there any performance improvement in using the first version over the second? Thank you Tommaso

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• jQueryUI autocomplete - when no results are returned

  - by Brian M. Hunt

  I'm wondering how one can catch and add a custom handler when empty results are returned from the server when using jQueryUI autocomplete. There seem to be a few questions on this point related to the various jQuery plugins (e.g. jQuery autocomplete display “No data” error message when results empty), but I am wondering if there's a better/simpler way to achieve the same with the jQueryUI autocomplete. It seems to me this is a common use case, and I thought perhaps that jQueryUI had improved on the jQuery autocomplete by adding the ability to cleanly handle this situation. However I've not been able to find documentation of such functionality, and before I hack away at it I'd like to throw out some feelers in case others have seen this before. While probably not particularly influential, I can have the server return anything - e.g. HTTP 204: No Content to a 200/JSON empty list - whatever makes it easiest to catch the result in jQueryUI's autocomplete. My first thought is to pass a callback with two arguments, namely a request object and a response callback to handle the code, per the documentation: The third variation, the callback, provides the most flexibility, and can be used to connect any data source to Autocomplete. The callback gets two arguments: A request object, with a single property called "term", which refers to the value currently in the text input. For example, when the user entered "new yo" in a city field, the Autocomplete term will equal "new yo". A response callback, which expects a single argument to contain the data to suggest to the user. This data should be filtered based on the provided term, and can be in any of the formats described above for simple local data (String-Array or Object-Array with label/value/both properties). When the response callback receives no data, it inserts returns a special one-line object-array that has a label and an indicator that there's no data (so the select/focus recognize it as the indicator that no-data was returned). This seems overcomplicated. I'd prefer to be able to use a source: "http://...", and just have a callback somewhere indicating that no data was returned. Thank you for reading. Brian

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• Color Theory: How to convert Munsell HVC to RGB/HSB/HSL

  - by Ian Boyd

  I'm looking at at document that describes the standard colors used in dentistry to describe the color of a tooth. They quote hue, value, chroma values, and indicate they are from the 1905 Munsell description of color: The system of colour notation developed by A. H. Munsell in 1905 identifies colour in terms of three attributes: HUE, VALUE (Brightness) and CHROMA (saturation) [15] HUE (H): Munsell defined hue as the quality by which we distinguish one colour from another. He selected five principle colours: red, yellow, green, blue, and purple; and five intermediate colours: yellow-red, green-yellow, blue-green, purple-blue, and red-purple. These were placed around a colour circle at equal points and the colours in between these points are a mixture of the two, in favour of the nearer point/colour (see Fig 1.). VALUE (V): This notation indicates the lightness or darkness of a colour in relation to a neutral grey scale, which extends from absolute black (value symbol 0) to absolute white (value symbol 10). This is essentially how ‘bright’ the colour is. CHROMA (C): This indicates the degree of divergence of a given hue from a neutral grey of the same value. The scale of chroma extends from 0 for a neutral grey to 10, 12, 14 or farther, depending upon the strength (saturation) of the sample to be evaluated. There are various systems for categorising colour, the Vita system is most commonly used in Dentistry. This uses the letters A, B, C and D to notate the hue (colour) of the tooth. The chroma and value are both indicated by a value from 1 to 4. A1 being lighter than A4, but A4 being more saturated than A1. If placed in order of value, i.e. brightness, the order from brightest to darkest would be: A1, B1, B2, A2, A3, D2, C1, B3, D3, D4, A3.5, B4, C2, A4, C3, C4 The exact values of Hue, Value and Chroma for each of the shades is shown below (16) So my question is, can anyone convert Munsell HVC into RGB, HSB or HSL? Hue Value (Brightness) Chroma(Saturation) === ================== ================== 4.5 7.80 1.7 2.4 7.45 2.6 1.3 7.40 2.9 1.6 7.05 3.2 1.6 6.70 3.1 5.1 7.75 1.6 4.3 7.50 2.2 2.3 7.25 3.2 2.4 7.00 3.2 4.3 7.30 1.6 2.8 6.90 2.3 2.6 6.70 2.3 1.6 6.30 2.9 3.0 7.35 1.8 1.8 7.10 2.3 3.7 7.05 2.4 They say that Value(Brightness) varies from 0..10, which is fine. So i take 7.05 to mean 70.5%. But what is Hue measured in? i'm used to hue being measured in degrees (0..360). But the values i see would all be red - when they should be more yellow, or brown. Finally, it says that Choma/Saturation can range from 0..10 ...or even higher - which makes it sound like an arbitrary scale. So can anyone convert Munsell HVC to HSB or HSL, or better yet, RGB?

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• problem with template inheritance

  - by amiad

  I'm trying to understand whay i get an error on this code: (the error is under g++ unix compiler. VS is compiling OK) template<class T> class A { public: T t; public: A(const T& t1) : t(t1) {} virtual void Print() const { cout<<*this<<endl;} friend ostream& operator<<(ostream& out, const A<T>& a) { out<<"I'm "<<typeid(a).name()<<endl; out<<"I hold "<<typeid(a.t).name()<<endl; out<<"The inner value is: "<<a.t<<endl; return out; } }; template<class T> class B : public A<T> { public: B(const T& t1) : A<T>(t1) {} const T& get() const { return t; } }; int main() { A<int> a(9); a.Print(); B<A<int> > b(a); b.Print(); (b.get()).Print(); return 0; } This code is giving the following error: main.cpp: In member function 'const T& B::get() const': main.cpp:23: error: 't' was not declared in this scope It did compiled when i changed the code of B to this: template<class T> class B : public A<T> { public: B(const T& t1) : A<T>(t1) {} const T& get() const { return A<T>::t; } }; I just cant understand what is the problem with the first code... It doesn't make sense that i really need to write "A::" every time...

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• Can't build full html table in QTextEdit with std::for_each...

  - by mosg

  Hi. Here is my code function: void ReportHistory::update(void) { ui.output->clear(); ui.output->setCurrentFont(QFont("Arial", 8, QFont::Normal)); QString title = "My Title"; QStringList headers = QString("Header1,Header2,Header3,Header4,Header5,Header6").split(","); QString html = QString( "<html>" \ "<head>" \ "<meta Content=\"Text/html; charset=Windows-1251\">" \ "<title>%1</title>" \ "</head>" \ "<body bgcolor=#ffffff link=#5000A0>" \ "<p>%1</p>" \ "<table border=1 cellspacing=0 cellpadding=2>" \ "<tr bgcolor=#f0f0f0>" ).arg(title); foreach (QString header, headers) { html.append(QString("<th>%1</th>").arg(header)); } html.append("</tr>"); struct Fill { QString html_; Analytics::NavHistory::History::value_type prev_; Fill(QString html) : html_(html) {} void operator ()(const Analytics::NavHistory::History::value_type& entry) { QStringList line = (QString( "%1|%2|%3|%4|%5|%6" ).arg(value1, 15) .arg(value2 ? ' ' : 'C', 8) .arg(value3, 15) .arg(value4, 15, 'f', 4) .arg(value5, 15) .arg(value6, 15, 'f', 4)).split("|"); html_.append("<tr>"); foreach (QString item, line) { html_.append("<td bkcolor=0>%1</td>").arg(item); } html_.append("</tr>"); prev_ = entry; } }; std::for_each(history_->data().begin(), history_->data().end(), Fill(html)); html.append( "</table>" \ "</body>" \ "</html>"); ui.output->setHtml(html); } Where: ui.output is a pointer to QTextEdit. Question: the ui.output just show me the headers, and not the full table, what is wrong? Thanks.

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• Haskell newbie on types

  - by garulfo

  I'm completely new to Haskell (and more generally to functional programming), so forgive me if this is really basic stuff. To get more than a taste, I try to implement in Haskell some algorithmic stuff I'm working on. I have a simple module Interval that implements intervals on the line. It contains the type data Interval t = Interval t t the helper function makeInterval :: (Ord t) => t -> t -> Interval t makeInterval l r | l <= r = Interval l r | otherwise = error "bad interval" and some utility functions about intervals. Here, my interest lies in multidimensional intervals (d-intervals), those objects that are composed of d intervals. I want to separately consider d-intervals that are the union of d disjoint intervals on the line (multiple interval) from those that are the union of d interval on d separate lines (track interval). With distinct algorithmic treatments in mind, I think it would be nice to have two distinct types (even if both are lists of intervals here) such as import qualified Interval as I -- Multilple interval newtype MInterval t = MInterval [I.Interval t] -- Track interval newtype TInterval t = TInterval [I.Interval t] to allow for distinct sanity checks, e.g. makeMInterval :: (Ord t) => [I.Interval t] -> MInterval t makeMInterval is = if foldr (&&) True [I.precedes i i' | (i, i') <- zip is (tail is)] then (MInterval is) else error "bad multiple interval" makeTInterval :: (Ord t) => [I.Interval t] -> TInterval t makeTInterval = TInterval I now get to the point, at last! But some functions are naturally concerned with both multiple intervals and track intervals. For example, a function order would return the number of intervals in a multiple interval or a track interval. What can I do? Adding -- Dimensional interval data DInterval t = MIntervalStuff (MInterval t) | TIntervalStuff (TInterval t) does not help much, since, if I understand well (correct me if I'm wrong), I would have to write order :: DInterval t -> Int order (MIntervalStuff (MInterval is)) = length is order (TIntervalStuff (TInterval is)) = length is and call order as order (MIntervalStuff is) or order (TIntervalStuff is) when is is a MInterval or a TInterval. Not that great, it looks odd. Neither I want to duplicate the function (I have many functions that are concerned with both multiple and track intevals, and some other d-interval definitions such as equal length multiple and track intervals). I'm left with the feeling that I'm completely wrong and have missed some important point about types in Haskell (and/or can't forget enough here about OO programming). So, quite a newbie question, what would be the best way in Haskell to deal with such a situation? Do I have to forget about introducing MInterval and TInterval and go with one type only? Thanks a lot for your help, Garulfo

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