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  • How to Reuse Your Old Wi-Fi Router as a Network Switch

    - by Jason Fitzpatrick
    Just because your old Wi-Fi router has been replaced by a newer model doesn’t mean it needs to gather dust in the closet. Read on as we show you how to take an old and underpowered Wi-Fi router and turn it into a respectable network switch (saving your $20 in the process). Image by mmgallan. Why Do I Want To Do This? Wi-Fi technology has changed significantly in the last ten years but Ethernet-based networking has changed very little. As such, a Wi-Fi router with 2006-era guts is lagging significantly behind current Wi-Fi router technology, but the Ethernet networking component of the device is just as useful as ever; aside from potentially being only 100Mbs instead of 1000Mbs capable (which for 99% of home applications is irrelevant) Ethernet is Ethernet. What does this matter to you, the consumer? It means that even though your old router doesn’t hack it for your Wi-Fi needs any longer the device is still a perfectly serviceable (and high quality) network switch. When do you need a network switch? Any time you want to share an Ethernet cable among multiple devices, you need a switch. For example, let’s say you have a single Ethernet wall jack behind your entertainment center. Unfortunately you have four devices that you want to link to your local network via hardline including your smart HDTV, DVR, Xbox, and a little Raspberry Pi running XBMC. Instead of spending $20-30 to purchase a brand new switch of comparable build quality to your old Wi-Fi router it makes financial sense (and is environmentally friendly) to invest five minutes of your time tweaking the settings on the old router to turn it from a Wi-Fi access point and routing tool into a network switch–perfect for dropping behind your entertainment center so that your DVR, Xbox, and media center computer can all share an Ethernet connection. What Do I Need? For this tutorial you’ll need a few things, all of which you likely have readily on hand or are free for download. To follow the basic portion of the tutorial, you’ll need the following: 1 Wi-Fi router with Ethernet ports 1 Computer with Ethernet jack 1 Ethernet cable For the advanced tutorial you’ll need all of those things, plus: 1 copy of DD-WRT firmware for your Wi-Fi router We’re conducting the experiment with a Linksys WRT54GL Wi-Fi router. The WRT54 series is one of the best selling Wi-Fi router series of all time and there’s a good chance a significant number of readers have one (or more) of them stuffed in an office closet. Even if you don’t have one of the WRT54 series routers, however, the principles we’re outlining here apply to all Wi-Fi routers; as long as your router administration panel allows the necessary changes you can follow right along with us. A quick note on the difference between the basic and advanced versions of this tutorial before we proceed. Your typical Wi-Fi router has 5 Ethernet ports on the back: 1 labeled “Internet”, “WAN”, or a variation thereof and intended to be connected to your DSL/Cable modem, and 4 labeled 1-4 intended to connect Ethernet devices like computers, printers, and game consoles directly to the Wi-Fi router. When you convert a Wi-Fi router to a switch, in most situations, you’ll lose two port as the “Internet” port cannot be used as a normal switch port and one of the switch ports becomes the input port for the Ethernet cable linking the switch to the main network. This means, referencing the diagram above, you’d lose the WAN port and LAN port 1, but retain LAN ports 2, 3, and 4 for use. If you only need to switch for 2-3 devices this may be satisfactory. However, for those of you that would prefer a more traditional switch setup where there is a dedicated WAN port and the rest of the ports are accessible, you’ll need to flash a third-party router firmware like the powerful DD-WRT onto your device. Doing so opens up the router to a greater degree of modification and allows you to assign the previously reserved WAN port to the switch, thus opening up LAN ports 1-4. Even if you don’t intend to use that extra port, DD-WRT offers you so many more options that it’s worth the extra few steps. Preparing Your Router for Life as a Switch Before we jump right in to shutting down the Wi-Fi functionality and repurposing your device as a network switch, there are a few important prep steps to attend to. First, you want to reset the router (if you just flashed a new firmware to your router, skip this step). Following the reset procedures for your particular router or go with what is known as the “Peacock Method” wherein you hold down the reset button for thirty seconds, unplug the router and wait (while still holding the reset button) for thirty seconds, and then plug it in while, again, continuing to hold down the rest button. Over the life of a router there are a variety of changes made, big and small, so it’s best to wipe them all back to the factory default before repurposing the router as a switch. Second, after resetting, we need to change the IP address of the device on the local network to an address which does not directly conflict with the new router. The typical default IP address for a home router is 192.168.1.1; if you ever need to get back into the administration panel of the router-turned-switch to check on things or make changes it will be a real hassle if the IP address of the device conflicts with the new home router. The simplest way to deal with this is to assign an address close to the actual router address but outside the range of addresses that your router will assign via the DHCP client; a good pick then is 192.168.1.2. Once the router is reset (or re-flashed) and has been assigned a new IP address, it’s time to configure it as a switch. Basic Router to Switch Configuration If you don’t want to (or need to) flash new firmware onto your device to open up that extra port, this is the section of the tutorial for you: we’ll cover how to take a stock router, our previously mentioned WRT54 series Linksys, and convert it to a switch. Hook the Wi-Fi router up to the network via one of the LAN ports (consider the WAN port as good as dead from this point forward, unless you start using the router in its traditional function again or later flash a more advanced firmware to the device, the port is officially retired at this point). Open the administration control panel via  web browser on a connected computer. Before we get started two things: first,  anything we don’t explicitly instruct you to change should be left in the default factory-reset setting as you find it, and two, change the settings in the order we list them as some settings can’t be changed after certain features are disabled. To start, let’s navigate to Setup ->Basic Setup. Here you need to change the following things: Local IP Address: [different than the primary router, e.g. 192.168.1.2] Subnet Mask: [same as the primary router, e.g. 255.255.255.0] DHCP Server: Disable Save with the “Save Settings” button and then navigate to Setup -> Advanced Routing: Operating Mode: Router This particular setting is very counterintuitive. The “Operating Mode” toggle tells the device whether or not it should enable the Network Address Translation (NAT)  feature. Because we’re turning a smart piece of networking hardware into a relatively dumb one, we don’t need this feature so we switch from Gateway mode (NAT on) to Router mode (NAT off). Our next stop is Wireless -> Basic Wireless Settings: Wireless SSID Broadcast: Disable Wireless Network Mode: Disabled After disabling the wireless we’re going to, again, do something counterintuitive. Navigate to Wireless -> Wireless Security and set the following parameters: Security Mode: WPA2 Personal WPA Algorithms: TKIP+AES WPA Shared Key: [select some random string of letters, numbers, and symbols like JF#d$di!Hdgio890] Now you may be asking yourself, why on Earth are we setting a rather secure Wi-Fi configuration on a Wi-Fi router we’re not going to use as a Wi-Fi node? On the off chance that something strange happens after, say, a power outage when your router-turned-switch cycles on and off a bunch of times and the Wi-Fi functionality is activated we don’t want to be running the Wi-Fi node wide open and granting unfettered access to your network. While the chances of this are next-to-nonexistent, it takes only a few seconds to apply the security measure so there’s little reason not to. Save your changes and navigate to Security ->Firewall. Uncheck everything but Filter Multicast Firewall Protect: Disable At this point you can save your changes again, review the changes you’ve made to ensure they all stuck, and then deploy your “new” switch wherever it is needed. Advanced Router to Switch Configuration For the advanced configuration, you’ll need a copy of DD-WRT installed on your router. Although doing so is an extra few steps, it gives you a lot more control over the process and liberates an extra port on the device. Hook the Wi-Fi router up to the network via one of the LAN ports (later you can switch the cable to the WAN port). Open the administration control panel via web browser on the connected computer. Navigate to the Setup -> Basic Setup tab to get started. In the Basic Setup tab, ensure the following settings are adjusted. The setting changes are not optional and are required to turn the Wi-Fi router into a switch. WAN Connection Type: Disabled Local IP Address: [different than the primary router, e.g. 192.168.1.2] Subnet Mask: [same as the primary router, e.g. 255.255.255.0] DHCP Server: Disable In addition to disabling the DHCP server, also uncheck all the DNSMasq boxes as the bottom of the DHCP sub-menu. If you want to activate the extra port (and why wouldn’t you), in the WAN port section: Assign WAN Port to Switch [X] At this point the router has become a switch and you have access to the WAN port so the LAN ports are all free. Since we’re already in the control panel, however, we might as well flip a few optional toggles that further lock down the switch and prevent something odd from happening. The optional settings are arranged via the menu you find them in. Remember to save your settings with the save button before moving onto a new tab. While still in the Setup -> Basic Setup menu, change the following: Gateway/Local DNS : [IP address of primary router, e.g. 192.168.1.1] NTP Client : Disable The next step is to turn off the radio completely (which not only kills the Wi-Fi but actually powers the physical radio chip off). Navigate to Wireless -> Advanced Settings -> Radio Time Restrictions: Radio Scheduling: Enable Select “Always Off” There’s no need to create a potential security problem by leaving the Wi-Fi radio on, the above toggle turns it completely off. Under Services -> Services: DNSMasq : Disable ttraff Daemon : Disable Under the Security -> Firewall tab, uncheck every box except “Filter Multicast”, as seen in the screenshot above, and then disable SPI Firewall. Once you’re done here save and move on to the Administration tab. Under Administration -> Management:  Info Site Password Protection : Enable Info Site MAC Masking : Disable CRON : Disable 802.1x : Disable Routing : Disable After this final round of tweaks, save and then apply your settings. Your router has now been, strategically, dumbed down enough to plod along as a very dependable little switch. Time to stuff it behind your desk or entertainment center and streamline your cabling.     

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  • Option Trading: Getting the most out of the event session options

    - by extended_events
    You can control different aspects of how an event session behaves by setting the event session options as part of the CREATE EVENT SESSION DDL. The default settings for the event session options are designed to handle most of the common event collection situations so I generally recommend that you just use the defaults. Like everything in the real world though, there are going to be a handful of “special cases” that require something different. This post focuses on identifying the special cases and the correct use of the options to accommodate those cases. There is a reason it’s called Default The default session options specify a total event buffer size of 4 MB with a 30 second latency. Translating this into human terms; this means that our default behavior is that the system will start processing events from the event buffer when we reach about 1.3 MB of events or after 30 seconds, which ever comes first. Aside: What’s up with the 1.3 MB, I thought you said the buffer was 4 MB?The Extended Events engine takes the total buffer size specified by MAX_MEMORY (4MB by default) and divides it into 3 equally sized buffers. This is done so that a session can be publishing events to one buffer while other buffers are being processed. There are always at least three buffers; how to get more than three is covered later. Using this configuration, the Extended Events engine can “keep up” with most event sessions on standard workloads. Why is this? The fact is that most events are small, really small; on the order of a couple hundred bytes. Even when you start considering events that carry dynamically sized data (eg. binary, text, etc.) or adding actions that collect additional data, the total size of the event is still likely to be pretty small. This means that each buffer can likely hold thousands of events before it has to be processed. When the event buffers are finally processed there is an economy of scale achieved since most targets support bulk processing of the events so they are processed at the buffer level rather than the individual event level. When all this is working together it’s more likely that a full buffer will be processed and put back into the ready queue before the remaining buffers (remember, there are at least three) are full. I know what you’re going to say: “My server is exceptional! My workload is so massive it defies categorization!” OK, maybe you weren’t going to say that exactly, but you were probably thinking it. The point is that there are situations that won’t be covered by the Default, but that’s a good place to start and this post assumes you’ve started there so that you have something to look at in order to determine if you do have a special case that needs different settings. So let’s get to the special cases… What event just fired?! How about now?! Now?! If you believe the commercial adage from Heinz Ketchup (Heinz Slow Good Ketchup ad on You Tube), some things are worth the wait. This is not a belief held by most DBAs, particularly DBAs who are looking for an answer to a troubleshooting question fast. If you’re one of these anxious DBAs, or maybe just a Program Manager doing a demo, then 30 seconds might be longer than you’re comfortable waiting. If you find yourself in this situation then consider changing the MAX_DISPATCH_LATENCY option for your event session. This option will force the event buffers to be processed based on your time schedule. This option only makes sense for the asynchronous targets since those are the ones where we allow events to build up in the event buffer – if you’re using one of the synchronous targets this option isn’t relevant. Avoid forgotten events by increasing your memory Have you ever had one of those days where you keep forgetting things? That can happen in Extended Events too; we call it dropped events. In order to optimizes for server performance and help ensure that the Extended Events doesn’t block the server if to drop events that can’t be published to a buffer because the buffer is full. You can determine if events are being dropped from a session by querying the dm_xe_sessions DMV and looking at the dropped_event_count field. Aside: Should you care if you’re dropping events?Maybe not – think about why you’re collecting data in the first place and whether you’re really going to miss a few dropped events. For example, if you’re collecting query duration stats over thousands of executions of a query it won’t make a huge difference to miss a couple executions. Use your best judgment. If you find that your session is dropping events it means that the event buffer is not large enough to handle the volume of events that are being published. There are two ways to address this problem. First, you could collect fewer events – examine you session to see if you are over collecting. Do you need all the actions you’ve specified? Could you apply a predicate to be more specific about when you fire the event? Assuming the session is defined correctly, the next option is to change the MAX_MEMORY option to a larger number. Picking the right event buffer size might take some trial and error, but a good place to start is with the number of dropped events compared to the number you’ve collected. Aside: There are three different behaviors for dropping events that you specify using the EVENT_RETENTION_MODE option. The default is to allow single event loss and you should stick with this setting since it is the best choice for keeping the impact on server performance low.You’ll be tempted to use the setting to not lose any events (NO_EVENT_LOSS) – resist this urge since it can result in blocking on the server. If you’re worried that you’re losing events you should be increasing your event buffer memory as described in this section. Some events are too big to fail A less common reason for dropping an event is when an event is so large that it can’t fit into the event buffer. Even though most events are going to be small, you might find a condition that occasionally generates a very large event. You can determine if your session is dropping large events by looking at the dm_xe_sessions DMV once again, this time check the largest_event_dropped_size. If this value is larger than the size of your event buffer [remember, the size of your event buffer, by default, is max_memory / 3] then you need a large event buffer. To specify a large event buffer you set the MAX_EVENT_SIZE option to a value large enough to fit the largest event dropped based on data from the DMV. When you set this option the Extended Events engine will create two buffers of this size to accommodate these large events. As an added bonus (no extra charge) the large event buffer will also be used to store normal events in the cases where the normal event buffers are all full and waiting to be processed. (Note: This is just a side-effect, not the intended use. If you’re dropping many normal events then you should increase your normal event buffer size.) Partitioning: moving your events to a sub-division Earlier I alluded to the fact that you can configure your event session to use more than the standard three event buffers – this is called partitioning and is controlled by the MEMORY_PARTITION_MODE option. The result of setting this option is fairly easy to explain, but knowing when to use it is a bit more art than science. First the science… You can configure partitioning in three ways: None, Per NUMA Node & Per CPU. This specifies the location where sets of event buffers are created with fairly obvious implication. There are rules we follow for sub-dividing the total memory (specified by MAX_MEMORY) between all the event buffers that are specific to the mode used: None: 3 buffers (fixed)Node: 3 * number_of_nodesCPU: 2.5 * number_of_cpus Here are some examples of what this means for different Node/CPU counts: Configuration None Node CPU 2 CPUs, 1 Node 3 buffers 3 buffers 5 buffers 6 CPUs, 2 Node 3 buffers 6 buffers 15 buffers 40 CPUs, 5 Nodes 3 buffers 15 buffers 100 buffers   Aside: Buffer size on multi-processor computersAs the number of Nodes or CPUs increases, the size of the event buffer gets smaller because the total memory is sub-divided into more pieces. The defaults will hold up to this for a while since each buffer set is holding events only from the Node or CPU that it is associated with, but at some point the buffers will get too small and you’ll either see events being dropped or you’ll get an error when you create your session because you’re below the minimum buffer size. Increase the MAX_MEMORY setting to an appropriate number for the configuration. The most likely reason to start partitioning is going to be related to performance. If you notice that running an event session is impacting the performance of your server beyond a reasonably expected level [Yes, there is a reasonably expected level of work required to collect events.] then partitioning might be an answer. Before you partition you might want to check a few other things: Is your event retention set to NO_EVENT_LOSS and causing blocking? (I told you not to do this.) Consider changing your event loss mode or increasing memory. Are you over collecting and causing more work than necessary? Consider adding predicates to events or removing unnecessary events and actions from your session. Are you writing the file target to the same slow disk that you use for TempDB and your other high activity databases? <kidding> <not really> It’s always worth considering the end to end picture – if you’re writing events to a file you can be impacted by I/O, network; all the usual stuff. Assuming you’ve ruled out the obvious (and not so obvious) issues, there are performance conditions that will be addressed by partitioning. For example, it’s possible to have a successful event session (eg. no dropped events) but still see a performance impact because you have many CPUs all attempting to write to the same free buffer and having to wait in line to finish their work. This is a case where partitioning would relieve the contention between the different CPUs and likely reduce the performance impact cause by the event session. There is no DMV you can check to find these conditions – sorry – that’s where the art comes in. This is  largely a matter of experimentation. On the bright side you probably won’t need to to worry about this level of detail all that often. The performance impact of Extended Events is significantly lower than what you may be used to with SQL Trace. You will likely only care about the impact if you are trying to set up a long running event session that will be part of your everyday workload – sessions used for short term troubleshooting will likely fall into the “reasonably expected impact” category. Hey buddy – I think you forgot something OK, there are two options I didn’t cover: STARTUP_STATE & TRACK_CAUSALITY. If you want your event sessions to start automatically when the server starts, set the STARTUP_STATE option to ON. (Now there is only one option I didn’t cover.) I’m going to leave causality for another post since it’s not really related to session behavior, it’s more about event analysis. - Mike Share this post: email it! | bookmark it! | digg it! | reddit! | kick it! | live it!

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  • Pluralsight Meet the Author Podcast on HTML5 Canvas Programming

    - by dwahlin
      In the latest installment of Pluralsight’s Meet the Author podcast series, Fritz Onion and I talk about my new course, HTML5 Canvas Fundamentals.  In the interview I describe different canvas technologies covered throughout the course and a sample application at the end of the course that covers how to build a custom business chart from start to finish. Meet the Author:  Dan Wahlin on HTML5 Canvas Fundamentals   Transcript [Fritz] Hi. This is Fritz Onion. I’m here today with Dan Wahlin to talk about his new course HTML5 Canvas Fundamentals. Dan founded the Wahlin Group, which you can find at thewahlingroup.com, which specializes in ASP.NET, jQuery, Silverlight, and SharePoint consulting. He’s a Microsoft Regional Director and has been awarded Microsoft’s MVP for ASP.NET, Connected Systems, and Silverlight. Dan is on the INETA Bureau’s — Speaker’s Bureau, speaks at conferences and user groups around the world, and has written several books on .NET. Thanks for talking to me today, Dan. [Dan] Always good to talk with you, Fritz. [Fritz] So this new course of yours, HTML5 Canvas Fundamentals, I have to say that most of the really snazzy demos I’ve seen with HTML5 have involved Canvas, so I thought it would be a good starting point to chat with you about why we decided to create a course dedicated just to Canvas. If you want to kind of give us that perspective. [Dan] Sure. So, you know, there’s quite a bit of material out there on HTML5 in general, and as people that have done a lot with HTML5 are probably aware, a lot of HTML5 is actually JavaScript centric. You know, a lot of people when they first learn it, think it’s tags, but most of it’s actually JavaScript, and it just so happens that the HTML5 Canvas is one of those things. And so it’s not just, you know, a tag you add and it just magically draws all these things. You mentioned there’s a lot of cool things you can do from games to there’s some really cool multimedia applications out there where they integrate video and audio and all kinds of things into the Canvas, to more business scenarios such as charting and things along those lines. So the reason we made a course specifically on it is, a lot of the material out there touches on it but the Canvas is actually a pretty deep topic. You can do some pretty advanced stuff or easy stuff depending on what your application requirements are, and the API itself, you know, there’s over 30 functions just in the Canvas API and then a whole set of properties that actually go with that as well. So it’s a pretty big topic, and that’s why we created a course specifically tailored towards just the Canvas. [Fritz] Right. And let’s — let me just review the outline briefly here for everyone. So you start off with an introduction to getting started with Canvas, drawing with the HTML5 Canvas, then you talk about manipulating pixels, and you finish up with building a custom data chart. So I really like your example flow here. I think it will appeal to even business developers, right. Even if you’re not into HTML5 for the games or the media capabilities, there’s still something here for everyone I think working with the Canvas. Which leads me to another question, which is, where do you see the Canvas fitting in to kind of your day-to-day developer, people that are working business applications and maybe vanilla websites that aren’t doing kind of cutting edge stuff with interactivity with users? Is there a still a place for the Canvas in those scenarios? [Dan] Yeah, definitely. I think a lot of us — and I include myself here — over the last few years, the focus has generally been, especially if you’re, let’s say, a PHP or ASP.NET or Java type of developer, we’re kind of accustomed to working on the server side, and, you know, we kind of relied on Flash or Silverlight or these other plug-ins for the client side stuff when it was kind of fancy, like charts and graphs and things along those lines. With the what I call massive shift of applications, you know, mainly because of mobile, to more of client side, one of the big benefits I think from a maybe corporate standard way of thinking of things, since we do a lot of work with different corporations, is that, number one, rather than having to have the plug-in, which of course isn’t going to work on iPad and some of these other devices out there that are pretty popular, you can now use a built-in technology that all the modern browsers support, and that includes things like Safari on the iPad and iPhone and the Android tablets and things like that with their browsers, and actually render some really sophisticated charts. Whether you do it by scratch or from scratch or, you know, get a third party type of library involved, it’s just JavaScript. So it downloads fast so it’s good from a performance perspective; and when it comes to what you can render, it’s extremely robust. You can do everything from, you know, your basic circles to polygons or polylines to really advanced gradients as well and even provide some interactivity and animations, and that’s some of the stuff I touch upon in the class. In fact, you mentioned the last part of the outline there is building a custom data chart and that’s kind of gears towards more of the, what I’d call enterprise or corporate type developer. [Fritz] Yeah, that makes sense. And it’s, you know, a lot of the demos I’ve seen with HTML5 focus on more the interactivity and kind of game side of things, but the Canvas is such a diverse element within HTML5 that I can see it being applicable pretty much anywhere. So why don’t we talk a little bit about some of the specifics of what you cover? You talk about drawing and then manipulating pixels. You want to kind of give us the different ways of working with the Canvas and what some of those APIs provide for you? [Dan] Sure. So going all the way back to the start of the outline, we actually started off by showing different demonstrations of the Canvas in action, and we show some fun stuff — multimedia apps and games and things like that — and then also some more business scenarios; and then once you see that, hopefully it kinds of piques your interest and you go, oh, wow, this is actually pretty phenomenal what you can do. So then we start you off with, so how to you actually draw things. Now, there are some libraries out there that will draw things like graphs, but if you want to customize those or just build something you have from scratch, you need to know the basics, such as, you know, how do you draw circles and lines and arcs and Bezier curves and all those fancy types of shapes that a given chart may have on it or that a game may have in it for that matter. So we start off by covering what I call the core API functions; how do you, for instance, fill a rectangle or convert that to a square by setting the height and the width; how do you draw arcs or different types of curves and there’s different types supported such as I mentioned Bezier curves or quadratic curves; and then we also talk about how do you integrate text into it. You might have some images already that are just regular bitmap type images that you want to integrate, you can do that with a Canvas. And you can even sync video into the Canvas, which actually opens up some pretty interesting possibilities for both business and I think just general multimedia apps. Once you kind of get those core functions down for the basic shapes that you need to be able to draw on any type of Canvas, then we go a little deeper into what are the pixels that are there to manipulate. And that’s one of the important things to understand about the HTML5 Canvas, scalable vector graphics is another thing you can use now in the modern browsers; it’s vector based. Canvas is pixel based. And so we talk about how to do gradients, how can you do transforms, you know, how do you scale things or rotate things, which is extremely useful for charts ’cause you might have text that, you know, flips up on its side for a y-axis or something like that. And you can even do direct pixel manipulation. So it’s really, really powerful. If you want to get down to the RGBA level, you can do that, and I show how to do that in the course, and then kind of wrap that section up with some animation fundamentals. [Fritz] Great. Yeah, that’s really powerful stuff for programmatically rendering data to clients and responding to user inputs. Look forward to seeing what everyone’s going to come up with building this stuff. So great. That’s — that’s HTML5 Canvas Fundamentals with Dan Wahlin. Thanks very much, Dan. [Dan] Thanks again. I appreciate it.

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  • PASS Summit Feedback

    - by Rob Farley
    PASS Feedback came in last week. I also saw my dentist for some fillings... At the PASS Summit this year, I delivered a couple of regular sessions and a Lightning Talk. People told me they enjoyed it, but when the rankings came out, they showed that I didn’t score particularly well. Brent Ozar was keen to discuss it with me. Brent: PASS speaker feedback is out. You did two sessions and a Lightning Talk. How did you go? Rob: Not so well actually, thanks for asking. Brent: Ha! Sorry. Of course you know that's why I wanted to discuss this with you. I was in one of your sessions at SQLBits in the UK a month before PASS, and I thought you rocked. You've got a really good and distinctive delivery style.  Then I noticed your talks were ranked in the bottom quarter of the Summit ratings and wanted to discuss it. Rob: Yeah, I know. You did ask me if we could do this...  I should explain – my presentation style is not the stereotypical IT conference one. I throw in jokes, and try to engage the audience thoroughly. I find many talks amazingly dry, and I guess I try to buck that trend. I also run training courses, and find that I get a lot of feedback from people thanking me for keeping things interesting. That said, I also get feedback criticising me for my style, and that’s basically what’s happened here. For the rest of this discussion, let’s focus on my talk about the Incredible Shrinking Execution Plan, which I considered to be my main talk. Brent: I thought that session title was the very best one at the entire Summit, and I had it on my recommended sessions list.  In four words, you managed to sum up the topic and your sense of humor.  I read that and immediately thought, "People need to be in this session," and then it didn't score well.  Tell me about your scores. Rob: The questions on the feedback form covered the usefulness of the information, the speaker’s presentation skills, their knowledge of the subject, how well the session was described, the amount of time allocated, and the quality of the presentation materials. Brent: Presentation materials? But you don’t do slides.  Did they rate your thong? Rob: No-one saw my flip-flops in this talk, Brent. I created a script in Management Studio, and published that afterwards, but I think people will have scored that question based on the lack of slides. I wasn’t expecting to do particularly well on that one. That was the only section that didn’t have 5/5 as the most popular score. Brent: See, that sucks, because cookbook-style scripts are often some of my favorites.  Adam Machanic's Service Broker workbench series helped me immensely when I was prepping for the MCM.  As an attendee, I'd rather have a commented script than a slide deck.  So how did you rank so low? Rob: When I look at the scores that you got (based on your blog post), you got very few scores below 3 – people that felt strong enough about your talk to post a negative score. In my scores, between 5% and 10% were below 3 (except on the question about whether I knew my stuff – I guess I came as knowledgeable). Brent: Wow – so quite a few people really didn’t like your talk then? Rob: Yeah. Mind you, based on the comments, some people really loved it. I’d like to think that there would be a certain portion of the room who may have rated the talk as one of the best of the conference. Some of my comments included “amazing!”, “Best presentation so far!”, “Wow, best session yet”, “fantastic” and “Outstanding!”. I think lots of talks can be “Great”, but not so many talks can be “Outstanding” without the word losing its meaning. One wrote “Pretty amazing presentation, considering it was completely extemporaneous.” Brent: Extemporaneous, eh? Rob: Yeah. I guess they don’t realise how much preparation goes into coming across as unprepared. In many ways it’s much easier to give a written speech than to deliver a presentation without slides as a prompt. Brent: That delivery style, the really relaxed, casual, college-professor approach was one of the things I really liked about your presentation at SQLbits.  As somebody who presents a lot, I "get" it - I know how hard it is to come off as relaxed and comfortable with your own material.  It's like improv done by jazz players and comedians - if you've never tried it, you don't realize how hard it is.  People also don't realize how hard it is to make a tough subject fun. Rob: Yeah well... There will be people writing comments on this post that say I wasn't trying to make the subject fun, and that I was making it all about me. Sometimes the style works, sometimes it doesn't. Most of the comments mentioned the fact that I tell jokes, some in a nice way, but some not so much (and it wasn't just a PASS thing - that's the mix of feedback I generally get). One comment at PASS was: “great stand up comedian - not what I'm looking for at pass”, and there were certainly a few that said “too many jokes”. I’m not trying to do stand-up – jokes are my way of engaging with the audience while I demonstrate some of the amazing things that the Query Optimizer can do if you write your queries the right way. Some people didn’t think it was technical enough, but I’ve also had some people tell me that the concepts I’m explaining are deep and profound. Brent: To me, that's a hallmark of a great explanation - when someone says, "But of course it has to work that way - how could it work any other way?  It seems so simple and logical."  Well, sure it does when it's explained correctly, but now pick up any number of thick SQL Server books and try to understand the Redundant Joins concept.  I guarantee it'll take more than 45 minutes. Rob: Some people in my audiences realise that, but definitely not everyone. There's only so much you can tell someone that something is profound. Generally it's something that they either have an epiphany on or not. I like to lull my audience into knowing what's going on, and do something that surprises them. Gain their trust, build a rapport, and then show them the deeper truth of what just happened. Brent: So you've learned your lesson about presentation scores, right?  From here on out, you're going to be dry, humorless, and all your presentations will consist of you reading bullet points off the screen. Rob: No Brent, I’m not. I'm also not going to suggest that most presentations at PASS are like that. No-one tries to present like that. There's a big space to occupy between what "dry and humourless" and me. My difference is to focus on the relationship I have with the crowd, rather than focussing on delivering the perfect session. I want to see people smiling and know they're relaxed. I think most presenters focus on the material, which is completely reasonable and safe. I remember once hearing someone talking about product creation. They talked about mediocrity. They said that one of the worst things that people can ever say about your product is that it’s “good”. What you want is for 10% of the world to love it enough to want to buy it. If 10% the world gave me a dollar, I’d have more money than I could ever use (assuming it wasn’t the SAME dollar they were giving me I guess). Brent: It's the Raving Fans theory.  It's better to have a small number of raving customers than a large number of almost-but-not-really customers who don't care that much about your product or service.  I know exactly how you feel - when I got survey feedback from my Quest video presentation when I was dressed up in a Richard Simmons costume, some of the attendees said I was unprofessional and distracting.  Some of the attendees couldn't get enough and Photoshopped all kinds of stuff into the screen captures.  On a whole, I probably didn't score that well, and I'm fine with that.  It sucks to look at the scores though - do those lower scores bother you? Rob: Of course they do. It hurts deeply. I open myself up and give presentations in a very personal way. All presenters do that, and we all feel the pain of negative feedback. I hate coming 146th & 162nd out of 185, but have to acknowledge that many sessions did worse still. Plus, once I feel the wounds have healed, I’ll be able to remember that there are people in the world that rave about my presentation style, and figure that people will hopefully talk about me. One day maybe those people that don’t like my presentation style will stay away and I might be able to score better. You don’t pay to hear country music if you prefer western... Lots of people find chili too spicy, but it’s still a popular food. Brent: But don’t you want to appeal to everyone? Rob: I do, but I don’t want to be lukewarm as in Revelation 3:16. I’d rather disgust and be discussed. Well, maybe not ‘disgust’, but I don’t want to conform. Conformity just isn’t the same any more. I’m not sure I’ve ever been one to do that. I try not to offend, but definitely like to be different. Brent: Count me among your raving fans, sir.  Where can we see you next? Rob: Considering I live in Adelaide in Australia, I’m not about to appear at anyone’s local SQL Saturday. I’m still trying to plan which events I’ll get to in 2011. I’ve submitted abstracts for TechEd North America, but won’t hold my breath. I’m also considering the SQLBits conferences in the UK in April, PASS in October, and I’m sure I’ll do some LiveMeeting presentations for user groups. Online, people download some of my recent SQLBits presentations at http://bit.ly/RFSarg and http://bit.ly/Simplification though. And they can download a 5-minute MP3 of my Lightning Talk at http://www.lobsterpot.com.au/files/Collation.mp3, in which I try to explain the idea behind collation, using thongs as an example. Brent: I was in the audience for http://bit.ly/RFSarg. That was a great presentation. Rob: Thanks, Brent. Now where’s my dollar?

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  • Option Trading: Getting the most out of the event session options

    - by extended_events
    You can control different aspects of how an event session behaves by setting the event session options as part of the CREATE EVENT SESSION DDL. The default settings for the event session options are designed to handle most of the common event collection situations so I generally recommend that you just use the defaults. Like everything in the real world though, there are going to be a handful of “special cases” that require something different. This post focuses on identifying the special cases and the correct use of the options to accommodate those cases. There is a reason it’s called Default The default session options specify a total event buffer size of 4 MB with a 30 second latency. Translating this into human terms; this means that our default behavior is that the system will start processing events from the event buffer when we reach about 1.3 MB of events or after 30 seconds, which ever comes first. Aside: What’s up with the 1.3 MB, I thought you said the buffer was 4 MB?The Extended Events engine takes the total buffer size specified by MAX_MEMORY (4MB by default) and divides it into 3 equally sized buffers. This is done so that a session can be publishing events to one buffer while other buffers are being processed. There are always at least three buffers; how to get more than three is covered later. Using this configuration, the Extended Events engine can “keep up” with most event sessions on standard workloads. Why is this? The fact is that most events are small, really small; on the order of a couple hundred bytes. Even when you start considering events that carry dynamically sized data (eg. binary, text, etc.) or adding actions that collect additional data, the total size of the event is still likely to be pretty small. This means that each buffer can likely hold thousands of events before it has to be processed. When the event buffers are finally processed there is an economy of scale achieved since most targets support bulk processing of the events so they are processed at the buffer level rather than the individual event level. When all this is working together it’s more likely that a full buffer will be processed and put back into the ready queue before the remaining buffers (remember, there are at least three) are full. I know what you’re going to say: “My server is exceptional! My workload is so massive it defies categorization!” OK, maybe you weren’t going to say that exactly, but you were probably thinking it. The point is that there are situations that won’t be covered by the Default, but that’s a good place to start and this post assumes you’ve started there so that you have something to look at in order to determine if you do have a special case that needs different settings. So let’s get to the special cases… What event just fired?! How about now?! Now?! If you believe the commercial adage from Heinz Ketchup (Heinz Slow Good Ketchup ad on You Tube), some things are worth the wait. This is not a belief held by most DBAs, particularly DBAs who are looking for an answer to a troubleshooting question fast. If you’re one of these anxious DBAs, or maybe just a Program Manager doing a demo, then 30 seconds might be longer than you’re comfortable waiting. If you find yourself in this situation then consider changing the MAX_DISPATCH_LATENCY option for your event session. This option will force the event buffers to be processed based on your time schedule. This option only makes sense for the asynchronous targets since those are the ones where we allow events to build up in the event buffer – if you’re using one of the synchronous targets this option isn’t relevant. Avoid forgotten events by increasing your memory Have you ever had one of those days where you keep forgetting things? That can happen in Extended Events too; we call it dropped events. In order to optimizes for server performance and help ensure that the Extended Events doesn’t block the server if to drop events that can’t be published to a buffer because the buffer is full. You can determine if events are being dropped from a session by querying the dm_xe_sessions DMV and looking at the dropped_event_count field. Aside: Should you care if you’re dropping events?Maybe not – think about why you’re collecting data in the first place and whether you’re really going to miss a few dropped events. For example, if you’re collecting query duration stats over thousands of executions of a query it won’t make a huge difference to miss a couple executions. Use your best judgment. If you find that your session is dropping events it means that the event buffer is not large enough to handle the volume of events that are being published. There are two ways to address this problem. First, you could collect fewer events – examine you session to see if you are over collecting. Do you need all the actions you’ve specified? Could you apply a predicate to be more specific about when you fire the event? Assuming the session is defined correctly, the next option is to change the MAX_MEMORY option to a larger number. Picking the right event buffer size might take some trial and error, but a good place to start is with the number of dropped events compared to the number you’ve collected. Aside: There are three different behaviors for dropping events that you specify using the EVENT_RETENTION_MODE option. The default is to allow single event loss and you should stick with this setting since it is the best choice for keeping the impact on server performance low.You’ll be tempted to use the setting to not lose any events (NO_EVENT_LOSS) – resist this urge since it can result in blocking on the server. If you’re worried that you’re losing events you should be increasing your event buffer memory as described in this section. Some events are too big to fail A less common reason for dropping an event is when an event is so large that it can’t fit into the event buffer. Even though most events are going to be small, you might find a condition that occasionally generates a very large event. You can determine if your session is dropping large events by looking at the dm_xe_sessions DMV once again, this time check the largest_event_dropped_size. If this value is larger than the size of your event buffer [remember, the size of your event buffer, by default, is max_memory / 3] then you need a large event buffer. To specify a large event buffer you set the MAX_EVENT_SIZE option to a value large enough to fit the largest event dropped based on data from the DMV. When you set this option the Extended Events engine will create two buffers of this size to accommodate these large events. As an added bonus (no extra charge) the large event buffer will also be used to store normal events in the cases where the normal event buffers are all full and waiting to be processed. (Note: This is just a side-effect, not the intended use. If you’re dropping many normal events then you should increase your normal event buffer size.) Partitioning: moving your events to a sub-division Earlier I alluded to the fact that you can configure your event session to use more than the standard three event buffers – this is called partitioning and is controlled by the MEMORY_PARTITION_MODE option. The result of setting this option is fairly easy to explain, but knowing when to use it is a bit more art than science. First the science… You can configure partitioning in three ways: None, Per NUMA Node & Per CPU. This specifies the location where sets of event buffers are created with fairly obvious implication. There are rules we follow for sub-dividing the total memory (specified by MAX_MEMORY) between all the event buffers that are specific to the mode used: None: 3 buffers (fixed)Node: 3 * number_of_nodesCPU: 2.5 * number_of_cpus Here are some examples of what this means for different Node/CPU counts: Configuration None Node CPU 2 CPUs, 1 Node 3 buffers 3 buffers 5 buffers 6 CPUs, 2 Node 3 buffers 6 buffers 15 buffers 40 CPUs, 5 Nodes 3 buffers 15 buffers 100 buffers   Aside: Buffer size on multi-processor computersAs the number of Nodes or CPUs increases, the size of the event buffer gets smaller because the total memory is sub-divided into more pieces. The defaults will hold up to this for a while since each buffer set is holding events only from the Node or CPU that it is associated with, but at some point the buffers will get too small and you’ll either see events being dropped or you’ll get an error when you create your session because you’re below the minimum buffer size. Increase the MAX_MEMORY setting to an appropriate number for the configuration. The most likely reason to start partitioning is going to be related to performance. If you notice that running an event session is impacting the performance of your server beyond a reasonably expected level [Yes, there is a reasonably expected level of work required to collect events.] then partitioning might be an answer. Before you partition you might want to check a few other things: Is your event retention set to NO_EVENT_LOSS and causing blocking? (I told you not to do this.) Consider changing your event loss mode or increasing memory. Are you over collecting and causing more work than necessary? Consider adding predicates to events or removing unnecessary events and actions from your session. Are you writing the file target to the same slow disk that you use for TempDB and your other high activity databases? <kidding> <not really> It’s always worth considering the end to end picture – if you’re writing events to a file you can be impacted by I/O, network; all the usual stuff. Assuming you’ve ruled out the obvious (and not so obvious) issues, there are performance conditions that will be addressed by partitioning. For example, it’s possible to have a successful event session (eg. no dropped events) but still see a performance impact because you have many CPUs all attempting to write to the same free buffer and having to wait in line to finish their work. This is a case where partitioning would relieve the contention between the different CPUs and likely reduce the performance impact cause by the event session. There is no DMV you can check to find these conditions – sorry – that’s where the art comes in. This is  largely a matter of experimentation. On the bright side you probably won’t need to to worry about this level of detail all that often. The performance impact of Extended Events is significantly lower than what you may be used to with SQL Trace. You will likely only care about the impact if you are trying to set up a long running event session that will be part of your everyday workload – sessions used for short term troubleshooting will likely fall into the “reasonably expected impact” category. Hey buddy – I think you forgot something OK, there are two options I didn’t cover: STARTUP_STATE & TRACK_CAUSALITY. If you want your event sessions to start automatically when the server starts, set the STARTUP_STATE option to ON. (Now there is only one option I didn’t cover.) I’m going to leave causality for another post since it’s not really related to session behavior, it’s more about event analysis. - Mike Share this post: email it! | bookmark it! | digg it! | reddit! | kick it! | live it!

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  • value types in the vm

    - by john.rose
    value types in the vm p.p1 {margin: 0.0px 0.0px 0.0px 0.0px; font: 14.0px Times} p.p2 {margin: 0.0px 0.0px 14.0px 0.0px; font: 14.0px Times} p.p3 {margin: 0.0px 0.0px 12.0px 0.0px; font: 14.0px Times} p.p4 {margin: 0.0px 0.0px 15.0px 0.0px; font: 14.0px Times} p.p5 {margin: 0.0px 0.0px 0.0px 0.0px; font: 14.0px Courier} p.p6 {margin: 0.0px 0.0px 0.0px 0.0px; font: 14.0px Courier; min-height: 17.0px} p.p7 {margin: 0.0px 0.0px 0.0px 0.0px; font: 14.0px Times; min-height: 18.0px} p.p8 {margin: 0.0px 0.0px 0.0px 36.0px; text-indent: -36.0px; font: 14.0px Times; min-height: 18.0px} p.p9 {margin: 0.0px 0.0px 12.0px 0.0px; font: 14.0px Times; min-height: 18.0px} p.p10 {margin: 0.0px 0.0px 12.0px 0.0px; font: 14.0px Times; color: #000000} li.li1 {margin: 0.0px 0.0px 0.0px 0.0px; font: 14.0px Times} li.li7 {margin: 0.0px 0.0px 0.0px 0.0px; font: 14.0px Times; min-height: 18.0px} span.s1 {font: 14.0px Courier} span.s2 {color: #000000} span.s3 {font: 14.0px Courier; color: #000000} ol.ol1 {list-style-type: decimal} Or, enduring values for a changing world. Introduction A value type is a data type which, generally speaking, is designed for being passed by value in and out of methods, and stored by value in data structures. The only value types which the Java language directly supports are the eight primitive types. Java indirectly and approximately supports value types, if they are implemented in terms of classes. For example, both Integer and String may be viewed as value types, especially if their usage is restricted to avoid operations appropriate to Object. In this note, we propose a definition of value types in terms of a design pattern for Java classes, accompanied by a set of usage restrictions. We also sketch the relation of such value types to tuple types (which are a JVM-level notion), and point out JVM optimizations that can apply to value types. This note is a thought experiment to extend the JVM’s performance model in support of value types. The demonstration has two phases.  Initially the extension can simply use design patterns, within the current bytecode architecture, and in today’s Java language. But if the performance model is to be realized in practice, it will probably require new JVM bytecode features, changes to the Java language, or both.  We will look at a few possibilities for these new features. An Axiom of Value In the context of the JVM, a value type is a data type equipped with construction, assignment, and equality operations, and a set of typed components, such that, whenever two variables of the value type produce equal corresponding values for their components, the values of the two variables cannot be distinguished by any JVM operation. Here are some corollaries: A value type is immutable, since otherwise a copy could be constructed and the original could be modified in one of its components, allowing the copies to be distinguished. Changing the component of a value type requires construction of a new value. The equals and hashCode operations are strictly component-wise. If a value type is represented by a JVM reference, that reference cannot be successfully synchronized on, and cannot be usefully compared for reference equality. A value type can be viewed in terms of what it doesn’t do. We can say that a value type omits all value-unsafe operations, which could violate the constraints on value types.  These operations, which are ordinarily allowed for Java object types, are pointer equality comparison (the acmp instruction), synchronization (the monitor instructions), all the wait and notify methods of class Object, and non-trivial finalize methods. The clone method is also value-unsafe, although for value types it could be treated as the identity function. Finally, and most importantly, any side effect on an object (however visible) also counts as an value-unsafe operation. A value type may have methods, but such methods must not change the components of the value. It is reasonable and useful to define methods like toString, equals, and hashCode on value types, and also methods which are specifically valuable to users of the value type. Representations of Value Value types have two natural representations in the JVM, unboxed and boxed. An unboxed value consists of the components, as simple variables. For example, the complex number x=(1+2i), in rectangular coordinate form, may be represented in unboxed form by the following pair of variables: /*Complex x = Complex.valueOf(1.0, 2.0):*/ double x_re = 1.0, x_im = 2.0; These variables might be locals, parameters, or fields. Their association as components of a single value is not defined to the JVM. Here is a sample computation which computes the norm of the difference between two complex numbers: double distance(/*Complex x:*/ double x_re, double x_im,         /*Complex y:*/ double y_re, double y_im) {     /*Complex z = x.minus(y):*/     double z_re = x_re - y_re, z_im = x_im - y_im;     /*return z.abs():*/     return Math.sqrt(z_re*z_re + z_im*z_im); } A boxed representation groups component values under a single object reference. The reference is to a ‘wrapper class’ that carries the component values in its fields. (A primitive type can naturally be equated with a trivial value type with just one component of that type. In that view, the wrapper class Integer can serve as a boxed representation of value type int.) The unboxed representation of complex numbers is practical for many uses, but it fails to cover several major use cases: return values, array elements, and generic APIs. The two components of a complex number cannot be directly returned from a Java function, since Java does not support multiple return values. The same story applies to array elements: Java has no ’array of structs’ feature. (Double-length arrays are a possible workaround for complex numbers, but not for value types with heterogeneous components.) By generic APIs I mean both those which use generic types, like Arrays.asList and those which have special case support for primitive types, like String.valueOf and PrintStream.println. Those APIs do not support unboxed values, and offer some problems to boxed values. Any ’real’ JVM type should have a story for returns, arrays, and API interoperability. The basic problem here is that value types fall between primitive types and object types. Value types are clearly more complex than primitive types, and object types are slightly too complicated. Objects are a little bit dangerous to use as value carriers, since object references can be compared for pointer equality, and can be synchronized on. Also, as many Java programmers have observed, there is often a performance cost to using wrapper objects, even on modern JVMs. Even so, wrapper classes are a good starting point for talking about value types. If there were a set of structural rules and restrictions which would prevent value-unsafe operations on value types, wrapper classes would provide a good notation for defining value types. This note attempts to define such rules and restrictions. Let’s Start Coding Now it is time to look at some real code. Here is a definition, written in Java, of a complex number value type. @ValueSafe public final class Complex implements java.io.Serializable {     // immutable component structure:     public final double re, im;     private Complex(double re, double im) {         this.re = re; this.im = im;     }     // interoperability methods:     public String toString() { return "Complex("+re+","+im+")"; }     public List<Double> asList() { return Arrays.asList(re, im); }     public boolean equals(Complex c) {         return re == c.re && im == c.im;     }     public boolean equals(@ValueSafe Object x) {         return x instanceof Complex && equals((Complex) x);     }     public int hashCode() {         return 31*Double.valueOf(re).hashCode()                 + Double.valueOf(im).hashCode();     }     // factory methods:     public static Complex valueOf(double re, double im) {         return new Complex(re, im);     }     public Complex changeRe(double re2) { return valueOf(re2, im); }     public Complex changeIm(double im2) { return valueOf(re, im2); }     public static Complex cast(@ValueSafe Object x) {         return x == null ? ZERO : (Complex) x;     }     // utility methods and constants:     public Complex plus(Complex c)  { return new Complex(re+c.re, im+c.im); }     public Complex minus(Complex c) { return new Complex(re-c.re, im-c.im); }     public double abs() { return Math.sqrt(re*re + im*im); }     public static final Complex PI = valueOf(Math.PI, 0.0);     public static final Complex ZERO = valueOf(0.0, 0.0); } This is not a minimal definition, because it includes some utility methods and other optional parts.  The essential elements are as follows: The class is marked as a value type with an annotation. The class is final, because it does not make sense to create subclasses of value types. The fields of the class are all non-private and final.  (I.e., the type is immutable and structurally transparent.) From the supertype Object, all public non-final methods are overridden. The constructor is private. Beyond these bare essentials, we can observe the following features in this example, which are likely to be typical of all value types: One or more factory methods are responsible for value creation, including a component-wise valueOf method. There are utility methods for complex arithmetic and instance creation, such as plus and changeIm. There are static utility constants, such as PI. The type is serializable, using the default mechanisms. There are methods for converting to and from dynamically typed references, such as asList and cast. The Rules In order to use value types properly, the programmer must avoid value-unsafe operations.  A helpful Java compiler should issue errors (or at least warnings) for code which provably applies value-unsafe operations, and should issue warnings for code which might be correct but does not provably avoid value-unsafe operations.  No such compilers exist today, but to simplify our account here, we will pretend that they do exist. A value-safe type is any class, interface, or type parameter marked with the @ValueSafe annotation, or any subtype of a value-safe type.  If a value-safe class is marked final, it is in fact a value type.  All other value-safe classes must be abstract.  The non-static fields of a value class must be non-public and final, and all its constructors must be private. Under the above rules, a standard interface could be helpful to define value types like Complex.  Here is an example: @ValueSafe public interface ValueType extends java.io.Serializable {     // All methods listed here must get redefined.     // Definitions must be value-safe, which means     // they may depend on component values only.     List<? extends Object> asList();     int hashCode();     boolean equals(@ValueSafe Object c);     String toString(); } //@ValueSafe inherited from supertype: public final class Complex implements ValueType { … The main advantage of such a conventional interface is that (unlike an annotation) it is reified in the runtime type system.  It could appear as an element type or parameter bound, for facilities which are designed to work on value types only.  More broadly, it might assist the JVM to perform dynamic enforcement of the rules for value types. Besides types, the annotation @ValueSafe can mark fields, parameters, local variables, and methods.  (This is redundant when the type is also value-safe, but may be useful when the type is Object or another supertype of a value type.)  Working forward from these annotations, an expression E is defined as value-safe if it satisfies one or more of the following: The type of E is a value-safe type. E names a field, parameter, or local variable whose declaration is marked @ValueSafe. E is a call to a method whose declaration is marked @ValueSafe. E is an assignment to a value-safe variable, field reference, or array reference. E is a cast to a value-safe type from a value-safe expression. E is a conditional expression E0 ? E1 : E2, and both E1 and E2 are value-safe. Assignments to value-safe expressions and initializations of value-safe names must take their values from value-safe expressions. A value-safe expression may not be the subject of a value-unsafe operation.  In particular, it cannot be synchronized on, nor can it be compared with the “==” operator, not even with a null or with another value-safe type. In a program where all of these rules are followed, no value-type value will be subject to a value-unsafe operation.  Thus, the prime axiom of value types will be satisfied, that no two value type will be distinguishable as long as their component values are equal. More Code To illustrate these rules, here are some usage examples for Complex: Complex pi = Complex.valueOf(Math.PI, 0); Complex zero = pi.changeRe(0);  //zero = pi; zero.re = 0; ValueType vtype = pi; @SuppressWarnings("value-unsafe")   Object obj = pi; @ValueSafe Object obj2 = pi; obj2 = new Object();  // ok List<Complex> clist = new ArrayList<Complex>(); clist.add(pi);  // (ok assuming List.add param is @ValueSafe) List<ValueType> vlist = new ArrayList<ValueType>(); vlist.add(pi);  // (ok) List<Object> olist = new ArrayList<Object>(); olist.add(pi);  // warning: "value-unsafe" boolean z = pi.equals(zero); boolean z1 = (pi == zero);  // error: reference comparison on value type boolean z2 = (pi == null);  // error: reference comparison on value type boolean z3 = (pi == obj2);  // error: reference comparison on value type synchronized (pi) { }  // error: synch of value, unpredictable result synchronized (obj2) { }  // unpredictable result Complex qq = pi; qq = null;  // possible NPE; warning: “null-unsafe" qq = (Complex) obj;  // warning: “null-unsafe" qq = Complex.cast(obj);  // OK @SuppressWarnings("null-unsafe")   Complex empty = null;  // possible NPE qq = empty;  // possible NPE (null pollution) The Payoffs It follows from this that either the JVM or the java compiler can replace boxed value-type values with unboxed ones, without affecting normal computations.  Fields and variables of value types can be split into their unboxed components.  Non-static methods on value types can be transformed into static methods which take the components as value parameters. Some common questions arise around this point in any discussion of value types. Why burden the programmer with all these extra rules?  Why not detect programs automagically and perform unboxing transparently?  The answer is that it is easy to break the rules accidently unless they are agreed to by the programmer and enforced.  Automatic unboxing optimizations are tantalizing but (so far) unreachable ideal.  In the current state of the art, it is possible exhibit benchmarks in which automatic unboxing provides the desired effects, but it is not possible to provide a JVM with a performance model that assures the programmer when unboxing will occur.  This is why I’m writing this note, to enlist help from, and provide assurances to, the programmer.  Basically, I’m shooting for a good set of user-supplied “pragmas” to frame the desired optimization. Again, the important thing is that the unboxing must be done reliably, or else programmers will have no reason to work with the extra complexity of the value-safety rules.  There must be a reasonably stable performance model, wherein using a value type has approximately the same performance characteristics as writing the unboxed components as separate Java variables. There are some rough corners to the present scheme.  Since Java fields and array elements are initialized to null, value-type computations which incorporate uninitialized variables can produce null pointer exceptions.  One workaround for this is to require such variables to be null-tested, and the result replaced with a suitable all-zero value of the value type.  That is what the “cast” method does above. Generically typed APIs like List<T> will continue to manipulate boxed values always, at least until we figure out how to do reification of generic type instances.  Use of such APIs will elicit warnings until their type parameters (and/or relevant members) are annotated or typed as value-safe.  Retrofitting List<T> is likely to expose flaws in the present scheme, which we will need to engineer around.  Here are a couple of first approaches: public interface java.util.List<@ValueSafe T> extends Collection<T> { … public interface java.util.List<T extends Object|ValueType> extends Collection<T> { … (The second approach would require disjunctive types, in which value-safety is “contagious” from the constituent types.) With more transformations, the return value types of methods can also be unboxed.  This may require significant bytecode-level transformations, and would work best in the presence of a bytecode representation for multiple value groups, which I have proposed elsewhere under the title “Tuples in the VM”. But for starters, the JVM can apply this transformation under the covers, to internally compiled methods.  This would give a way to express multiple return values and structured return values, which is a significant pain-point for Java programmers, especially those who work with low-level structure types favored by modern vector and graphics processors.  The lack of multiple return values has a strong distorting effect on many Java APIs. Even if the JVM fails to unbox a value, there is still potential benefit to the value type.  Clustered computing systems something have copy operations (serialization or something similar) which apply implicitly to command operands.  When copying JVM objects, it is extremely helpful to know when an object’s identity is important or not.  If an object reference is a copied operand, the system may have to create a proxy handle which points back to the original object, so that side effects are visible.  Proxies must be managed carefully, and this can be expensive.  On the other hand, value types are exactly those types which a JVM can “copy and forget” with no downside. Array types are crucial to bulk data interfaces.  (As data sizes and rates increase, bulk data becomes more important than scalar data, so arrays are definitely accompanying us into the future of computing.)  Value types are very helpful for adding structure to bulk data, so a successful value type mechanism will make it easier for us to express richer forms of bulk data. Unboxing arrays (i.e., arrays containing unboxed values) will provide better cache and memory density, and more direct data movement within clustered or heterogeneous computing systems.  They require the deepest transformations, relative to today’s JVM.  There is an impedance mismatch between value-type arrays and Java’s covariant array typing, so compromises will need to be struck with existing Java semantics.  It is probably worth the effort, since arrays of unboxed value types are inherently more memory-efficient than standard Java arrays, which rely on dependent pointer chains. It may be sufficient to extend the “value-safe” concept to array declarations, and allow low-level transformations to change value-safe array declarations from the standard boxed form into an unboxed tuple-based form.  Such value-safe arrays would not be convertible to Object[] arrays.  Certain connection points, such as Arrays.copyOf and System.arraycopy might need additional input/output combinations, to allow smooth conversion between arrays with boxed and unboxed elements. Alternatively, the correct solution may have to wait until we have enough reification of generic types, and enough operator overloading, to enable an overhaul of Java arrays. Implicit Method Definitions The example of class Complex above may be unattractively complex.  I believe most or all of the elements of the example class are required by the logic of value types. If this is true, a programmer who writes a value type will have to write lots of error-prone boilerplate code.  On the other hand, I think nearly all of the code (except for the domain-specific parts like plus and minus) can be implicitly generated. Java has a rule for implicitly defining a class’s constructor, if no it defines no constructors explicitly.  Likewise, there are rules for providing default access modifiers for interface members.  Because of the highly regular structure of value types, it might be reasonable to perform similar implicit transformations on value types.  Here’s an example of a “highly implicit” definition of a complex number type: public class Complex implements ValueType {  // implicitly final     public double re, im;  // implicitly public final     //implicit methods are defined elementwise from te fields:     //  toString, asList, equals(2), hashCode, valueOf, cast     //optionally, explicit methods (plus, abs, etc.) would go here } In other words, with the right defaults, a simple value type definition can be a one-liner.  The observant reader will have noticed the similarities (and suitable differences) between the explicit methods above and the corresponding methods for List<T>. Another way to abbreviate such a class would be to make an annotation the primary trigger of the functionality, and to add the interface(s) implicitly: public @ValueType class Complex { … // implicitly final, implements ValueType (But to me it seems better to communicate the “magic” via an interface, even if it is rooted in an annotation.) Implicitly Defined Value Types So far we have been working with nominal value types, which is to say that the sequence of typed components is associated with a name and additional methods that convey the intention of the programmer.  A simple ordered pair of floating point numbers can be variously interpreted as (to name a few possibilities) a rectangular or polar complex number or Cartesian point.  The name and the methods convey the intended meaning. But what if we need a truly simple ordered pair of floating point numbers, without any further conceptual baggage?  Perhaps we are writing a method (like “divideAndRemainder”) which naturally returns a pair of numbers instead of a single number.  Wrapping the pair of numbers in a nominal type (like “QuotientAndRemainder”) makes as little sense as wrapping a single return value in a nominal type (like “Quotient”).  What we need here are structural value types commonly known as tuples. For the present discussion, let us assign a conventional, JVM-friendly name to tuples, roughly as follows: public class java.lang.tuple.$DD extends java.lang.tuple.Tuple {      double $1, $2; } Here the component names are fixed and all the required methods are defined implicitly.  The supertype is an abstract class which has suitable shared declarations.  The name itself mentions a JVM-style method parameter descriptor, which may be “cracked” to determine the number and types of the component fields. The odd thing about such a tuple type (and structural types in general) is it must be instantiated lazily, in response to linkage requests from one or more classes that need it.  The JVM and/or its class loaders must be prepared to spin a tuple type on demand, given a simple name reference, $xyz, where the xyz is cracked into a series of component types.  (Specifics of naming and name mangling need some tasteful engineering.) Tuples also seem to demand, even more than nominal types, some support from the language.  (This is probably because notations for non-nominal types work best as combinations of punctuation and type names, rather than named constructors like Function3 or Tuple2.)  At a minimum, languages with tuples usually (I think) have some sort of simple bracket notation for creating tuples, and a corresponding pattern-matching syntax (or “destructuring bind”) for taking tuples apart, at least when they are parameter lists.  Designing such a syntax is no simple thing, because it ought to play well with nominal value types, and also with pre-existing Java features, such as method parameter lists, implicit conversions, generic types, and reflection.  That is a task for another day. Other Use Cases Besides complex numbers and simple tuples there are many use cases for value types.  Many tuple-like types have natural value-type representations. These include rational numbers, point locations and pixel colors, and various kinds of dates and addresses. Other types have a variable-length ‘tail’ of internal values. The most common example of this is String, which is (mathematically) a sequence of UTF-16 character values. Similarly, bit vectors, multiple-precision numbers, and polynomials are composed of sequences of values. Such types include, in their representation, a reference to a variable-sized data structure (often an array) which (somehow) represents the sequence of values. The value type may also include ’header’ information. Variable-sized values often have a length distribution which favors short lengths. In that case, the design of the value type can make the first few values in the sequence be direct ’header’ fields of the value type. In the common case where the header is enough to represent the whole value, the tail can be a shared null value, or even just a null reference. Note that the tail need not be an immutable object, as long as the header type encapsulates it well enough. This is the case with String, where the tail is a mutable (but never mutated) character array. Field types and their order must be a globally visible part of the API.  The structure of the value type must be transparent enough to have a globally consistent unboxed representation, so that all callers and callees agree about the type and order of components  that appear as parameters, return types, and array elements.  This is a trade-off between efficiency and encapsulation, which is forced on us when we remove an indirection enjoyed by boxed representations.  A JVM-only transformation would not care about such visibility, but a bytecode transformation would need to take care that (say) the components of complex numbers would not get swapped after a redefinition of Complex and a partial recompile.  Perhaps constant pool references to value types need to declare the field order as assumed by each API user. This brings up the delicate status of private fields in a value type.  It must always be possible to load, store, and copy value types as coordinated groups, and the JVM performs those movements by moving individual scalar values between locals and stack.  If a component field is not public, what is to prevent hostile code from plucking it out of the tuple using a rogue aload or astore instruction?  Nothing but the verifier, so we may need to give it more smarts, so that it treats value types as inseparable groups of stack slots or locals (something like long or double). My initial thought was to make the fields always public, which would make the security problem moot.  But public is not always the right answer; consider the case of String, where the underlying mutable character array must be encapsulated to prevent security holes.  I believe we can win back both sides of the tradeoff, by training the verifier never to split up the components in an unboxed value.  Just as the verifier encapsulates the two halves of a 64-bit primitive, it can encapsulate the the header and body of an unboxed String, so that no code other than that of class String itself can take apart the values. Similar to String, we could build an efficient multi-precision decimal type along these lines: public final class DecimalValue extends ValueType {     protected final long header;     protected private final BigInteger digits;     public DecimalValue valueOf(int value, int scale) {         assert(scale >= 0);         return new DecimalValue(((long)value << 32) + scale, null);     }     public DecimalValue valueOf(long value, int scale) {         if (value == (int) value)             return valueOf((int)value, scale);         return new DecimalValue(-scale, new BigInteger(value));     } } Values of this type would be passed between methods as two machine words. Small values (those with a significand which fits into 32 bits) would be represented without any heap data at all, unless the DecimalValue itself were boxed. (Note the tension between encapsulation and unboxing in this case.  It would be better if the header and digits fields were private, but depending on where the unboxing information must “leak”, it is probably safer to make a public revelation of the internal structure.) Note that, although an array of Complex can be faked with a double-length array of double, there is no easy way to fake an array of unboxed DecimalValues.  (Either an array of boxed values or a transposed pair of homogeneous arrays would be reasonable fallbacks, in a current JVM.)  Getting the full benefit of unboxing and arrays will require some new JVM magic. Although the JVM emphasizes portability, system dependent code will benefit from using machine-level types larger than 64 bits.  For example, the back end of a linear algebra package might benefit from value types like Float4 which map to stock vector types.  This is probably only worthwhile if the unboxing arrays can be packed with such values. More Daydreams A more finely-divided design for dynamic enforcement of value safety could feature separate marker interfaces for each invariant.  An empty marker interface Unsynchronizable could cause suitable exceptions for monitor instructions on objects in marked classes.  More radically, a Interchangeable marker interface could cause JVM primitives that are sensitive to object identity to raise exceptions; the strangest result would be that the acmp instruction would have to be specified as raising an exception. @ValueSafe public interface ValueType extends java.io.Serializable,         Unsynchronizable, Interchangeable { … public class Complex implements ValueType {     // inherits Serializable, Unsynchronizable, Interchangeable, @ValueSafe     … It seems possible that Integer and the other wrapper types could be retro-fitted as value-safe types.  This is a major change, since wrapper objects would be unsynchronizable and their references interchangeable.  It is likely that code which violates value-safety for wrapper types exists but is uncommon.  It is less plausible to retro-fit String, since the prominent operation String.intern is often used with value-unsafe code. We should also reconsider the distinction between boxed and unboxed values in code.  The design presented above obscures that distinction.  As another thought experiment, we could imagine making a first class distinction in the type system between boxed and unboxed representations.  Since only primitive types are named with a lower-case initial letter, we could define that the capitalized version of a value type name always refers to the boxed representation, while the initial lower-case variant always refers to boxed.  For example: complex pi = complex.valueOf(Math.PI, 0); Complex boxPi = pi;  // convert to boxed myList.add(boxPi); complex z = myList.get(0);  // unbox Such a convention could perhaps absorb the current difference between int and Integer, double and Double. It might also allow the programmer to express a helpful distinction among array types. As said above, array types are crucial to bulk data interfaces, but are limited in the JVM.  Extending arrays beyond the present limitations is worth thinking about; for example, the Maxine JVM implementation has a hybrid object/array type.  Something like this which can also accommodate value type components seems worthwhile.  On the other hand, does it make sense for value types to contain short arrays?  And why should random-access arrays be the end of our design process, when bulk data is often sequentially accessed, and it might make sense to have heterogeneous streams of data as the natural “jumbo” data structure.  These considerations must wait for another day and another note. More Work It seems to me that a good sequence for introducing such value types would be as follows: Add the value-safety restrictions to an experimental version of javac. Code some sample applications with value types, including Complex and DecimalValue. Create an experimental JVM which internally unboxes value types but does not require new bytecodes to do so.  Ensure the feasibility of the performance model for the sample applications. Add tuple-like bytecodes (with or without generic type reification) to a major revision of the JVM, and teach the Java compiler to switch in the new bytecodes without code changes. A staggered roll-out like this would decouple language changes from bytecode changes, which is always a convenient thing. A similar investigation should be applied (concurrently) to array types.  In this case, it seems to me that the starting point is in the JVM: Add an experimental unboxing array data structure to a production JVM, perhaps along the lines of Maxine hybrids.  No bytecode or language support is required at first; everything can be done with encapsulated unsafe operations and/or method handles. Create an experimental JVM which internally unboxes value types but does not require new bytecodes to do so.  Ensure the feasibility of the performance model for the sample applications. Add tuple-like bytecodes (with or without generic type reification) to a major revision of the JVM, and teach the Java compiler to switch in the new bytecodes without code changes. That’s enough musing me for now.  Back to work!

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  • Routing to a Controller with no View in Angular

    - by Rick Strahl
    I've finally had some time to put Angular to use this week in a small project I'm working on for fun. Angular's routing is great and makes it real easy to map URL routes to controllers and model data into views. But what if you don't actually need a view, if you effectively need a headless controller that just runs code, but doesn't render a view?Preserve the ViewWhen Angular navigates a route and and presents a new view, it loads the controller and then renders the view from scratch. Views are not cached or stored, but displayed and then removed. So if you have routes configured like this:'use strict'; // Declare app level module which depends on filters, and services window.myApp = angular.module('myApp', ['myApp.filters', 'myApp.services', 'myApp.directives', 'myApp.controllers']). config(['$routeProvider', function($routeProvider) { $routeProvider.when('/map', { template: "partials/map.html ", controller: 'mapController', reloadOnSearch: false, animation: 'slide' }); … $routeProvider.otherwise({redirectTo: '/map'}); }]); Angular routes to the mapController and then re-renders the map.html template with the new data from the $scope filled in.But, but… I don't want a new View!Now in most cases this works just fine. If I'm rendering plain DOM content, or textboxes in a form interface that is all fine and dandy - it's perfectly fine to completely re-render the UI.But in some cases, the UI that's being managed has state and shouldn't be redrawn. In this case the main page in question has a Google Map on it. The map is  going to be manipulated throughout the lifetime of the application and the rest of the pages. In my application I have a toolbar on the bottom and the rest of the content is replaced/switched out by the Angular Views:The problem is that the map shouldn't be redrawn each time the Location view is activated. It should maintain its state, such as the current position selected (which can move), and shouldn't redraw due to the overhead of re-rendering the initial map.Originally I set up the map, exactly like all my other views - as a partial, that is rendered with a separate file, but that didn't work.The Workaround - Controller Only RoutesThe workaround for this goes decidedly against Angular's way of doing things:Setting up a Template-less RouteIn-lining the map view directly into the main pageHiding and showing the map view manuallyLet's see how this works.Controller Only RouteThe template-less route is basically a route that doesn't have any template to render. This is not directly supported by Angular, but thankfully easy to fake. The end goal here is that I want to simply have the Controller fire and then have the controller manage the display of the already active view by hiding and showing the map and any other view content, in effect bypassing Angular's view display management.In short - I want a controller action, but no view rendering.The controller-only or template-less route looks like this: $routeProvider.when('/map', { template: " ", // just fire controller controller: 'mapController', animation: 'slide' });Notice I'm using the template property rather than templateUrl (used in the first example above), which allows specifying a string template, and leaving it blank. The template property basically allows you to provide a templated string using Angular's HandleBar like binding syntax which can be useful at times. You can use plain strings or strings with template code in the template, or as I'm doing here a blank string to essentially fake 'just clear the view'. In-lined ViewSo if there's no view where does the HTML go? Because I don't want Angular to manage the view the map markup is in-lined directly into the page. So instead of rendering the map into the Angular view container, the content is simply set up as inline HTML to display as a sibling to the view container.<div id="MapContent" data-icon="LocationIcon" ng-controller="mapController" style="display:none"> <div class="headerbar"> <div class="right-header" style="float:right"> <a id="btnShowSaveLocationDialog" class="iconbutton btn btn-sm" href="#/saveLocation" style="margin-right: 2px;"> <i class="icon-ok icon-2x" style="color: lightgreen; "></i> Save Location </a> </div> <div class="left-header">GeoCrumbs</div> </div> <div class="clearfix"></div> <div id="Message"> <i id="MessageIcon"></i> <span id="MessageText"></span> </div> <div id="Map" class="content-area"> </div> </div> <div id="ViewPlaceholder" ng-view></div>Note that there's the #MapContent element and the #ViewPlaceHolder. The #MapContent is my static map view that is always 'live' and is initially hidden. It is initially hidden and doesn't get made visible until the MapController controller activates it which does the initial rendering of the map. After that the element is persisted with the map data already loaded and any future access only updates the map with new locations/pins etc.Note that default route is assigned to the mapController, which means that the mapController is fired right as the page loads, which is actually a good thing in this case, as the map is the cornerstone of this app that is manipulated by some of the other controllers/views.The Controller handles some UISince there's effectively no view activation with the template-less route, the controller unfortunately has to take over some UI interaction directly. Specifically it has to swap the hidden state between the map and any of the other views.Here's what the controller looks like:myApp.controller('mapController', ["$scope", "$routeParams", "locationData", function($scope, $routeParams, locationData) { $scope.locationData = locationData.location; $scope.locationHistory = locationData.locationHistory; if ($routeParams.mode == "currentLocation") { bc.getCurrentLocation(false); } bc.showMap(false,"#LocationIcon"); }]);bc.showMap is responsible for a couple of display tasks that hide/show the views/map and for activating/deactivating icons. The code looks like this:this.showMap = function (hide,selActiveIcon) { if (!hide) $("#MapContent").show(); else { $("#MapContent").hide(); } self.fitContent(); if (selActiveIcon) { $(".iconbutton").removeClass("active"); $(selActiveIcon).addClass("active"); } };Each of the other controllers in the app also call this function when they are activated to basically hide the map and make the View Content area visible. The map controller makes the map.This is UI code and calling this sort of thing from controllers is generally not recommended, but I couldn't figure out a way using directives to make this work any more easily than this. It'd be easy to hide and show the map and view container using a flag an ng-show, but it gets tricky because of scoping of the $scope. I would have to resort to storing this setting on the $rootscope which I try to avoid. The same issues exists with the icons.It sure would be nice if Angular had a way to explicitly specify that a View shouldn't be destroyed when another view is activated, so currently this workaround is required. Searching around, I saw a number of whacky hacks to get around this, but this solution I'm using here seems much easier than any of that I could dig up even if it doesn't quite fit the 'Angular way'.Angular nice, until it's notOverall I really like Angular and the way it works although it took me a bit of time to get my head around how all the pieces fit together. Once I got the idea how the app/routes, the controllers and views snap together, putting together Angular pages becomes fairly straightforward. You can get quite a bit done never going beyond those basics. For most common things Angular's default routing and view presentation works very well.But, when you do something a bit more complex, where there are multiple dependencies or as in this case where Angular doesn't appear to support a feature that's absolutely necessary, you're on your own. Finding information on more advanced topics is not trivial especially since versions are changing so rapidly and the low level behaviors are changing frequently so finding something that works is often an exercise in trial and error. Not that this is surprising. Angular is a complex piece of kit as are all the frameworks that try to hack JavaScript into submission to do something that it was really never designed to. After all everything about a framework like Angular is an elaborate hack. A lot of shit has to happen to make this all work together and at that Angular (and Ember, Durandel etc.) are pretty amazing pieces of JavaScript code. So no harm, no foul, but I just can't help feeling like working in toy sandbox at times :-)© Rick Strahl, West Wind Technologies, 2005-2013Posted in Angular  JavaScript   Tweet !function(d,s,id){var js,fjs=d.getElementsByTagName(s)[0];if(!d.getElementById(id)){js=d.createElement(s);js.id=id;js.src="//platform.twitter.com/widgets.js";fjs.parentNode.insertBefore(js,fjs);}}(document,"script","twitter-wjs"); (function() { var po = document.createElement('script'); po.type = 'text/javascript'; po.async = true; po.src = 'https://apis.google.com/js/plusone.js'; var s = document.getElementsByTagName('script')[0]; s.parentNode.insertBefore(po, s); })();

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  • That Escalated Quickly

    - by Jesse Taber
    Originally posted on: http://geekswithblogs.net/GruffCode/archive/2014/05/17/that-escalated-quickly.aspxI have been working remotely out of my home for over 4 years now. All of my coworkers during that time have also worked remotely. Lots of folks have written about the challenges inherent in facilitating communication on remote teams and strategies for overcoming them. A popular theme around this topic is the notion of “escalating communication”. In this context “escalating” means taking a conversation from one mode of communication to a different, higher fidelity mode of communication. Here are the five modes of communication I use at work in order of increasing fidelity: Email – This is the “lowest fidelity” mode of communication that I use. I usually only check it a few times a day (and I’m trying to check it even less frequently than that) and I only keep items in my inbox if they represent an item I need to take action on that I haven’t tracked anywhere else. Forums / Message boards – Being a developer, I’ve gotten into the habit of having other people look over my code before it becomes part of the product I’m working on. These code reviews often happen in “real time” via screen sharing, but I also always have someone else give all of the changes another look using pull requests. A pull request takes my code and lets someone else see the changes I’ve made side-by-side with the existing code so they can see if I did anything dumb. Pull requests can facilitate a conversation about the code changes in an online-forum like style. Some teams I’ve worked on also liked using tools like Trello or Google Groups to have on-going conversations about a topic or task that was being worked on. Chat & Instant Messaging  - Chat and instant messaging are the real workhorses for communication on the remote teams I’ve been a part of. I know some teams that are co-located that also use it pretty extensively for quick messages that don’t warrant walking across the office to talk with someone but reqire more immediacy than an e-mail. For the purposes of this post I think it’s important to note that the terms “chat” and “instant messaging” might insinuate that the conversation is happening in real time, but that’s not always true. Modern chat and IM applications maintain a searchable history so people can easily see what might have been discussed while they were away from their computers. Voice, Video and Screen sharing – Everyone’s got a camera and microphone on their computers now, and there are an abundance of services that will let you use them to talk to other people who have cameras and microphones on their computers. I’m including screen sharing here as well because, in my experience, these discussions typically involve one or more people showing the other participants something that’s happening on their screen. Obviously, this mode of communication is much higher-fidelity than any of the ones listed above. Scheduled meetings are typically conducted using this mode of communication. In Person – No matter how great communication tools become, there’s no substitute for meeting with someone face-to-face. However, opportunities for this kind of communcation are few and far between when you work on a remote team. When a conversation gets escalated that usually means it moves up one or more positions on this list. A lot of people advocate jumping to #4 sooner than later. Like them, I used to believe that, if it was possible, organizing a call with voice and video was automatically better than any kind of text-based communication could be. Lately, however, I’m becoming less convinced that escalating is always the right move. Working Asynchronously Last year I attended a talk at our local code camp given by Drew Miller. Drew works at GitHub and was talking about how they use GitHub internally. Many of the folks at GitHub work remotely, so communication was one of the main themes in Drew’s talk. During the talk Drew used the phrase, “asynchronous communication” to describe their use of chat and pull request comments. That phrase stuck in my head because I hadn’t heard it before but I think it perfectly describes the way in which remote teams often need to communicate. You don’t always know when your co-workers are at their computers or what hours (if any) they are working that day. In order to work this way you need to assume that the person you’re talking to might not respond right away. You can’t always afford to wait until everyone required is online and available to join a voice call, so you need to use text-based, persistent forms of communication so that people can receive and respond to messages when they are available. Going back to my list from the beginning of this post for a second, I characterize items #1-3 as being “asynchronous” modes of communication while we could call items #4 and #5 “synchronous”. When communication gets escalated it’s almost always moving from an asynchronous mode of communication to a synchronous one. Now, to the point of this post: I’ve become increasingly reluctant to escalate from asynchronous to synchronous communication for two primary reasons: 1 – You can often find a higher fidelity way to convey your message without holding a synchronous conversation 2 - Asynchronous modes of communication are (usually) persistent and searchable. You Don’t Have to Broadcast Live Let’s start with the first reason I’ve listed. A lot of times you feel like you need to escalate to synchronous communication because you’re having difficulty describing something that you’re seeing in words. You want to provide the people you’re conversing with some audio-visual aids to help them understand the point that you’re trying to make and you think that getting on Skype and sharing your screen with them is the best way to do that. Firing up a screen sharing session does work well, but you can usually accomplish the same thing in an asynchronous manner. For example, you could take a screenshot and annotate it with some text and drawings to illustrate what it is you’re seeing. If a screenshot won’t work, taking a short screen recording while your narrate over it and posting the video to your forum or chat system along with a text-based description of what’s in the recording that can be searched for later can be a great way to effectively communicate with your team asynchronously. I Said What?!? Now for the second reason I listed: most asynchronous modes of communication provide a transcript of what was said and what decisions might have been made during the conversation. There have been many occasions where I’ve used the search feature of my team’s chat application to find a conversation that happened several weeks or months ago to remember what was decided. Unfortunately, I think the benefits associated with the persistence of communicating asynchronously often get overlooked when people decide to escalate to a in-person meeting or voice/video call. I’m becoming much more reluctant to suggest a voice or video call if I suspect that it might lead to codifying some kind of design decision because everyone involved is going to hang up the call and immediately forget what was decided. I recognize that you can record and archive these types of interactions, but without being able to search them the recordings aren’t terribly useful. When and How To Escalate I don’t mean to imply that communicating via voice/video or in person is never a good idea. I probably jump on a Skype call with a co-worker at least once a day to quickly hash something out or show them a bit of code that I’m working on. Also, meeting in person periodically is really important for remote teams. There’s no way around the fact that sometimes it’s easier to jump on a call and show someone my screen so they can see what I’m seeing. So when is it right to escalate? I think the simplest way to answer that is when the communication starts to feel painful. Everyone’s tolerance for that pain is different, but I think you need to let it hurt a little bit before jumping to synchronous communication. When you do escalate from asynchronous to synchronous communication, there are a couple of things you can do to maximize the effectiveness of the communication: Takes notes – This is huge and yet I’ve found that a lot of teams don’t do this. If you’re holding a meeting with  > 2 people you should have someone taking notes. Taking notes while participating in a meeting can be difficult but there are a few strategies to deal with this challenge that probably deserve a short post of their own. After the meeting, make sure the notes are posted to a place where all concerned parties (including those that might not have attended the meeting) can review and search them. Persist decisions made ASAP – If any decisions were made during the meeting, persist those decisions to a searchable medium as soon as possible following the conversation. All the teams I’ve worked on used a web-based system for tracking the on-going work and a backlog of work to be done in the future. I always try to make sure that all of the cards/stories/tasks/whatever in these systems always reflect the latest decisions that were made as the work was being planned and executed. If held a quick call with your team lead and decided that it wasn’t worth the effort to build real-time validation into that new UI you were working on, go and codify that decision in the story associated with that work immediately after you hang up. Even better, write it up in the story while you are both still on the phone. That way when the folks from your QA team pick up the story to test a few days later they’ll know why the real-time validation isn’t there without having to invoke yet another conversation about the work. Communicating Well is Hard At this point you might be thinking that communicating asynchronously is more difficult than having a live conversation. You’re right: it is more difficult. In order to communicate effectively this way you need to very carefully think about the message that you’re trying to convey and craft it in a way that’s easy for your audience to understand. This is almost always harder than just talking through a problem in real time with someone; this is why escalating communication is such a popular idea. Why wouldn’t we want to do the thing that’s easier? Easier isn’t always better. If you and your team can get in the habit of communicating effectively in an asynchronous manner you’ll find that, over time, all of your communications get less painful because you don’t need to re-iterate previously made points over and over again. If you communicate right the first time, you often don’t need to rehash old conversations because you can go back and find the decisions that were made laid out in plain language. You’ll also find that you get better at doing things like writing useful comments in your code, creating written documentation about how the feature that you just built works, or persuading your team to do things in a certain way.

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  • C#/.NET Little Wonders: Fun With Enum Methods

    - by James Michael Hare
    Once again lets dive into the Little Wonders of .NET, those small things in the .NET languages and BCL classes that make development easier by increasing readability, maintainability, and/or performance. So probably every one of us has used an enumerated type at one time or another in a C# program.  The enumerated types we create are a great way to represent that a value can be one of a set of discrete values (or a combination of those values in the case of bit flags). But the power of enum types go far beyond simple assignment and comparison, there are many methods in the Enum class (that all enum types “inherit” from) that can give you even more power when dealing with them. IsDefined() – check if a given value exists in the enum Are you reading a value for an enum from a data source, but are unsure if it is actually a valid value or not?  Casting won’t tell you this, and Parse() isn’t guaranteed to balk either if you give it an int or a combination of flags.  So what can we do? Let’s assume we have a small enum like this for result codes we want to return back from our business logic layer: 1: public enum ResultCode 2: { 3: Success, 4: Warning, 5: Error 6: } In this enum, Success will be zero (unless given another value explicitly), Warning will be one, and Error will be two. So what happens if we have code like this where perhaps we’re getting the result code from another data source (could be database, could be web service, etc)? 1: public ResultCode PerformAction() 2: { 3: // set up and call some method that returns an int. 4: int result = ResultCodeFromDataSource(); 5:  6: // this will suceed even if result is < 0 or > 2. 7: return (ResultCode) result; 8: } So what happens if result is –1 or 4?  Well, the cast does not fail, so what we end up with would be an instance of a ResultCode that would have a value that’s outside of the bounds of the enum constants we defined. This means if you had a block of code like: 1: switch (result) 2: { 3: case ResultType.Success: 4: // do success stuff 5: break; 6:  7: case ResultType.Warning: 8: // do warning stuff 9: break; 10:  11: case ResultType.Error: 12: // do error stuff 13: break; 14: } That you would hit none of these blocks (which is a good argument for always having a default in a switch by the way). So what can you do?  Well, there is a handy static method called IsDefined() on the Enum class which will tell you if an enum value is defined.  1: public ResultCode PerformAction() 2: { 3: int result = ResultCodeFromDataSource(); 4:  5: if (!Enum.IsDefined(typeof(ResultCode), result)) 6: { 7: throw new InvalidOperationException("Enum out of range."); 8: } 9:  10: return (ResultCode) result; 11: } In fact, this is often recommended after you Parse() or cast a value to an enum as there are ways for values to get past these methods that may not be defined. If you don’t like the syntax of passing in the type of the enum, you could clean it up a bit by creating an extension method instead that would allow you to call IsDefined() off any isntance of the enum: 1: public static class EnumExtensions 2: { 3: // helper method that tells you if an enum value is defined for it's enumeration 4: public static bool IsDefined(this Enum value) 5: { 6: return Enum.IsDefined(value.GetType(), value); 7: } 8: }   HasFlag() – an easier way to see if a bit (or bits) are set Most of us who came from the land of C programming have had to deal extensively with bit flags many times in our lives.  As such, using bit flags may be almost second nature (for a quick refresher on bit flags in enum types see one of my old posts here). However, in higher-level languages like C#, the need to manipulate individual bit flags is somewhat diminished, and the code to check for bit flag enum values may be obvious to an advanced developer but cryptic to a novice developer. For example, let’s say you have an enum for a messaging platform that contains bit flags: 1: // usually, we pluralize flags enum type names 2: [Flags] 3: public enum MessagingOptions 4: { 5: None = 0, 6: Buffered = 0x01, 7: Persistent = 0x02, 8: Durable = 0x04, 9: Broadcast = 0x08 10: } We can combine these bit flags using the bitwise OR operator (the ‘|’ pipe character): 1: // combine bit flags using 2: var myMessenger = new Messenger(MessagingOptions.Buffered | MessagingOptions.Broadcast); Now, if we wanted to check the flags, we’d have to test then using the bit-wise AND operator (the ‘&’ character): 1: if ((options & MessagingOptions.Buffered) == MessagingOptions.Buffered) 2: { 3: // do code to set up buffering... 4: // ... 5: } While the ‘|’ for combining flags is easy enough to read for advanced developers, the ‘&’ test tends to be easy for novice developers to get wrong.  First of all you have to AND the flag combination with the value, and then typically you should test against the flag combination itself (and not just for a non-zero)!  This is because the flag combination you are testing with may combine multiple bits, in which case if only one bit is set, the result will be non-zero but not necessarily all desired bits! Thanks goodness in .NET 4.0 they gave us the HasFlag() method.  This method can be called from an enum instance to test to see if a flag is set, and best of all you can avoid writing the bit wise logic yourself.  Not to mention it will be more readable to a novice developer as well: 1: if (options.HasFlag(MessagingOptions.Buffered)) 2: { 3: // do code to set up buffering... 4: // ... 5: } It is much more concise and unambiguous, thus increasing your maintainability and readability. It would be nice to have a corresponding SetFlag() method, but unfortunately generic types don’t allow you to specialize on Enum, which makes it a bit more difficult.  It can be done but you have to do some conversions to numeric and then back to the enum which makes it less of a payoff than having the HasFlag() method.  But if you want to create it for symmetry, it would look something like this: 1: public static T SetFlag<T>(this Enum value, T flags) 2: { 3: if (!value.GetType().IsEquivalentTo(typeof(T))) 4: { 5: throw new ArgumentException("Enum value and flags types don't match."); 6: } 7:  8: // yes this is ugly, but unfortunately we need to use an intermediate boxing cast 9: return (T)Enum.ToObject(typeof (T), Convert.ToUInt64(value) | Convert.ToUInt64(flags)); 10: } Note that since the enum types are value types, we need to assign the result to something (much like string.Trim()).  Also, you could chain several SetFlag() operations together or create one that takes a variable arg list if desired. Parse() and ToString() – transitioning from string to enum and back Sometimes, you may want to be able to parse an enum from a string or convert it to a string - Enum has methods built in to let you do this.  Now, many may already know this, but may not appreciate how much power are in these two methods. For example, if you want to parse a string as an enum, it’s easy and works just like you’d expect from the numeric types: 1: string optionsString = "Persistent"; 2:  3: // can use Enum.Parse, which throws if finds something it doesn't like... 4: var result = (MessagingOptions)Enum.Parse(typeof (MessagingOptions), optionsString); 5:  6: if (result == MessagingOptions.Persistent) 7: { 8: Console.WriteLine("It worked!"); 9: } Note that Enum.Parse() will throw if it finds a value it doesn’t like.  But the values it likes are fairly flexible!  You can pass in a single value, or a comma separated list of values for flags and it will parse them all and set all bits: 1: // for string values, can have one, or comma separated. 2: string optionsString = "Persistent, Buffered"; 3:  4: var result = (MessagingOptions)Enum.Parse(typeof (MessagingOptions), optionsString); 5:  6: if (result.HasFlag(MessagingOptions.Persistent) && result.HasFlag(MessagingOptions.Buffered)) 7: { 8: Console.WriteLine("It worked!"); 9: } Or you can parse in a string containing a number that represents a single value or combination of values to set: 1: // 3 is the combination of Buffered (0x01) and Persistent (0x02) 2: var optionsString = "3"; 3:  4: var result = (MessagingOptions) Enum.Parse(typeof (MessagingOptions), optionsString); 5:  6: if (result.HasFlag(MessagingOptions.Persistent) && result.HasFlag(MessagingOptions.Buffered)) 7: { 8: Console.WriteLine("It worked again!"); 9: } And, if you really aren’t sure if the parse will work, and don’t want to handle an exception, you can use TryParse() instead: 1: string optionsString = "Persistent, Buffered"; 2: MessagingOptions result; 3:  4: // try parse returns true if successful, and takes an out parm for the result 5: if (Enum.TryParse(optionsString, out result)) 6: { 7: if (result.HasFlag(MessagingOptions.Persistent) && result.HasFlag(MessagingOptions.Buffered)) 8: { 9: Console.WriteLine("It worked!"); 10: } 11: } So we covered parsing a string to an enum, what about reversing that and converting an enum to a string?  The ToString() method is the obvious and most basic choice for most of us, but did you know you can pass a format string for enum types that dictate how they are written as a string?: 1: MessagingOptions value = MessagingOptions.Buffered | MessagingOptions.Persistent; 2:  3: // general format, which is the default, 4: Console.WriteLine("Default : " + value); 5: Console.WriteLine("G (default): " + value.ToString("G")); 6:  7: // Flags format, even if type does not have Flags attribute. 8: Console.WriteLine("F (flags) : " + value.ToString("F")); 9:  10: // integer format, value as number. 11: Console.WriteLine("D (num) : " + value.ToString("D")); 12:  13: // hex format, value as hex 14: Console.WriteLine("X (hex) : " + value.ToString("X")); Which displays: 1: Default : Buffered, Persistent 2: G (default): Buffered, Persistent 3: F (flags) : Buffered, Persistent 4: D (num) : 3 5: X (hex) : 00000003 Now, you may not really see a difference here between G and F because I used a [Flags] enum, the difference is that the “F” option treats the enum as if it were flags even if the [Flags] attribute is not present.  Let’s take a non-flags enum like the ResultCode used earlier: 1: // yes, we can do this even if it is not [Flags] enum. 2: ResultCode value = ResultCode.Warning | ResultCode.Error; And if we run that through the same formats again we get: 1: Default : 3 2: G (default): 3 3: F (flags) : Warning, Error 4: D (num) : 3 5: X (hex) : 00000003 Notice that since we had multiple values combined, but it was not a [Flags] marked enum, the G and default format gave us a number instead of a value name.  This is because the value was not a valid single-value constant of the enum.  However, using the F flags format string, it broke out the value into its component flags even though it wasn’t marked [Flags]. So, if you want to get an enum to display appropriately for whether or not it has the [Flags] attribute, use G which is the default.  If you always want it to attempt to break down the flags, use F.  For numeric output, obviously D or  X are the best choice depending on whether you want decimal or hex. Summary Hopefully, you learned a couple of new tricks with using the Enum class today!  I’ll add more little wonders as I think of them and thanks for all the invaluable input!   Technorati Tags: C#,.NET,Little Wonders,Enum,BlackRabbitCoder

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  • Making a Case For The Command Line

    - by Jesse Taber
    Originally posted on: http://geekswithblogs.net/GruffCode/archive/2013/06/30/making-a-case-for-the-command-line.aspxI have had an idea percolating in the back of my mind for over a year now that I’ve just recently started to implement. This idea relates to building out “internal tools” to ease the maintenance and on-going support of a software system. The system that I currently work on is (mostly) web-based, so we traditionally we have built these internal tools in the form of pages within the app that are only accessible by our developers and support personnel. These pages allow us to perform tasks within the system that, for one reason or another, we don’t want to let our end users perform (e.g. mass create/update/delete operations on data, flipping switches that turn paid modules of the system on or off, etc). When we try to build new tools like this we often struggle with the level of effort required to build them. Effort Required Creating a whole new page in an existing web application can be a fairly large undertaking. You need to create the page and ensure it will have a layout that is consistent with the other pages in the app. You need to decide what types of input controls need to go onto the page. You need to ensure that everything uses the same style as the rest of the site. You need to figure out what the text on the page should say. Then, when you figure out that you forgot about an input that should really be present you might have to go back and re-work the entire thing. Oh, and in addition to all of that, you still have to, you know, write the code that actually performs the task. Everything other than the code that performs the task at hand is just overhead. We don’t need a fancy date picker control in a nicely styled page for the vast majority of our internal tools. We don’t even really need a page, for that matter. We just need a way to issue a command to the application and have it, in turn, execute the code that we’ve written to accomplish a given task. All we really need is a simple console application! Plumbing Problems A former co-worker of mine, John Sonmez, always advocated the Unix philosophy for building internal tools: start with something that runs at the command line, and then build a UI on top of that if you need to. John’s idea has a lot of merit, and we tried building out some internal tools as simple Console applications. Unfortunately, this was often easier said that done. Doing a “File –> New Project” to build out a tool for a mature system can be pretty daunting because that new project is totally empty.  In our case, the web application code had a lot of of “plumbing” built in: it managed authentication and authorization, it handled database connection management for our multi-tenanted architecture, it managed all of the context that needs to follow a user around the application such as their timezone and regional/language settings. In addition, the configuration file for the web application  (a web.config in our case because this is an ASP .NET application) is large and would need to be reproduced into a similar configuration file for a Console application. While most of these problems are could be solved pretty easily with some refactoring of the codebase, building Console applications for internal tools still potentially suffers from one pretty big drawback: you’d have to execute them on a machine with network access to all of the needed resources. Obviously, our web servers can easily communicate the the database servers and can publish messages to our service bus, but the same is not true for all of our developer and support personnel workstations. We could have everyone run these tools remotely via RDP or SSH, but that’s a bit cumbersome and certainly a lot less convenient than having the tools built into the web application that is so easily accessible. Mix and Match So we need a way to build tools that are easily accessible via the web application but also don’t require the overhead of creating a user interface. This is where my idea comes into play: why not just build a command line interface into the web application? If it’s part of the web application we get all of the plumbing that comes along with that code, and we’re executing everything on the web servers which means we’ll have access to any external resources that we might need. Rather than having to incur the overhead of creating a brand new page for each tool that we want to build, we can create one new page that simply accepts a command in text form and executes it as a request on the web server. In this way, we can focus on writing the code to accomplish the task. If the tool ends up being heavily used, then (and only then) should we consider spending the time to build a better user experience around it. To be clear, I’m not trying to downplay the importance of building great user experiences into your system; we should all strive to provide the best UX possible to our end users. I’m only advocating this sort of bare-bones interface for internal consumption by the technical staff that builds and supports the software. This command line interface should be the “back end” to a highly polished and eye-pleasing public face. Implementation As I mentioned at the beginning of this post, this is an idea that I’ve had for awhile but have only recently started building out. I’ve outlined some general guidelines and design goals for this effort as follows: Text in, text out: In the interest of keeping things as simple as possible, I want this interface to be purely text-based. Users will submit commands as plain text, and the application will provide responses in plain text. Obviously this text will be “wrapped” within the context of HTTP requests and responses, but I don’t want to have to think about HTML or CSS when taking input from the user or displaying responses back to the user. Task-oriented code only: After building the initial “harness” for this interface, the only code that should need to be written to create a new internal tool should be code that is expressly needed to accomplish the task that the tool is intended to support. If we want to encourage and enable ourselves to build good tooling, we need to lower the barriers to entry as much as possible. Built-in documentation: One of the great things about most command line utilities is the ‘help’ switch that provides usage guidelines and details about the arguments that the utility accepts. Our web-based command line utility should allow us to build the documentation for these tools directly into the code of the tools themselves. I finally started trying to implement this idea when I heard about a fantastic open-source library called CLAP (Command Line Auto Parser) that lets me meet the guidelines outlined above. CLAP lets you define classes with public methods that can be easily invoked from the command line. Here’s a quick example of the code that would be needed to create a new tool to do something within your system: 1: public class CustomerTools 2: { 3: [Verb] 4: public void UpdateName(int customerId, string firstName, string lastName) 5: { 6: //invoke internal services/domain objects/hwatever to perform update 7: } 8: } This is just a regular class with a single public method (though you could have as many methods as you want). The method is decorated with the ‘Verb’ attribute that tells the CLAP library that it is a method that can be invoked from the command line. Here is how you would invoke that code: Parser.Run(args, new CustomerTools()); Note that ‘args’ is just a string[] that would normally be passed passed in from the static Main method of a Console application. Also, CLAP allows you to pass in multiple classes that define [Verb] methods so you can opt to organize the code that CLAP will invoke in any way that you like. You can invoke this code from a command line application like this: SomeExe UpdateName -customerId:123 -firstName:Jesse -lastName:Taber ‘SomeExe’ in this example just represents the name of .exe that is would be created from our Console application. CLAP then interprets the arguments passed in order to find the method that should be invoked and automatically parses out the parameters that need to be passed in. After a quick spike, I’ve found that invoking the ‘Parser’ class can be done from within the context of a web application just as easily as it can from within the ‘Main’ method entry point of a Console application. There are, however, a few sticking points that I’m working around: Splitting arguments into the ‘args’ array like the command line: When you invoke a standard .NET console application you get the arguments that were passed in by the user split into a handy array (this is the ‘args’ parameter referenced above). Generally speaking they get split by whitespace, but it’s also clever enough to handle things like ignoring whitespace in a phrase that is surrounded by quotes. We’ll need to re-create this logic within our web application so that we can give the ‘args’ value to CLAP just like a console application would. Providing a response to the user: If you were writing a console application, you might just use Console.WriteLine to provide responses to the user as to the progress and eventual outcome of the command. We can’t use Console.WriteLine within a web application, so I’ll need to find another way to provide feedback to the user. Preferably this approach would allow me to use the same handler classes from both a Console application and a web application, so some kind of strategy pattern will likely emerge from this effort. Submitting files: Often an internal tool needs to support doing some kind of operation in bulk, and the easiest way to submit the data needed to support the bulk operation is in a file. Getting the file uploaded and available to the CLAP handler classes will take a little bit of effort. Mimicking the console experience: This isn’t really a requirement so much as a “nice to have”. To start out, the command-line interface in the web application will probably be a single ‘textarea’ control with a button to submit the contents to a handler that will pass it along to CLAP to be parsed and run. I think it would be interesting to use some javascript and CSS trickery to change that page into something with more of a “shell” interface look and feel. I’ll be blogging more about this effort in the future and will include some code snippets (or maybe even a full blown example app) as I progress. I also think that I’ll probably end up either submitting some pull requests to the CLAP project or possibly forking/wrapping it into a more web-friendly package and open sourcing that.

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  • 256 Windows Azure Worker Roles, Windows Kinect and a 90's Text-Based Ray-Tracer

    - by Alan Smith
    For a couple of years I have been demoing a simple render farm hosted in Windows Azure using worker roles and the Azure Storage service. At the start of the presentation I deploy an Azure application that uses 16 worker roles to render a 1,500 frame 3D ray-traced animation. At the end of the presentation, when the animation was complete, I would play the animation delete the Azure deployment. The standing joke with the audience was that it was that it was a “$2 demo”, as the compute charges for running the 16 instances for an hour was $1.92, factor in the bandwidth charges and it’s a couple of dollars. The point of the demo is that it highlights one of the great benefits of cloud computing, you pay for what you use, and if you need massive compute power for a short period of time using Windows Azure can work out very cost effective. The “$2 demo” was great for presenting at user groups and conferences in that it could be deployed to Azure, used to render an animation, and then removed in a one hour session. I have always had the idea of doing something a bit more impressive with the demo, and scaling it from a “$2 demo” to a “$30 demo”. The challenge was to create a visually appealing animation in high definition format and keep the demo time down to one hour.  This article will take a run through how I achieved this. Ray Tracing Ray tracing, a technique for generating high quality photorealistic images, gained popularity in the 90’s with companies like Pixar creating feature length computer animations, and also the emergence of shareware text-based ray tracers that could run on a home PC. In order to render a ray traced image, the ray of light that would pass from the view point must be tracked until it intersects with an object. At the intersection, the color, reflectiveness, transparency, and refractive index of the object are used to calculate if the ray will be reflected or refracted. Each pixel may require thousands of calculations to determine what color it will be in the rendered image. Pin-Board Toys Having very little artistic talent and a basic understanding of maths I decided to focus on an animation that could be modeled fairly easily and would look visually impressive. I’ve always liked the pin-board desktop toys that become popular in the 80’s and when I was working as a 3D animator back in the 90’s I always had the idea of creating a 3D ray-traced animation of a pin-board, but never found the energy to do it. Even if I had a go at it, the render time to produce an animation that would look respectable on a 486 would have been measured in months. PolyRay Back in 1995 I landed my first real job, after spending three years being a beach-ski-climbing-paragliding-bum, and was employed to create 3D ray-traced animations for a CD-ROM that school kids would use to learn physics. I had got into the strange and wonderful world of text-based ray tracing, and was using a shareware ray-tracer called PolyRay. PolyRay takes a text file describing a scene as input and, after a few hours processing on a 486, produced a high quality ray-traced image. The following is an example of a basic PolyRay scene file. background Midnight_Blue   static define matte surface { ambient 0.1 diffuse 0.7 } define matte_white texture { matte { color white } } define matte_black texture { matte { color dark_slate_gray } } define position_cylindrical 3 define lookup_sawtooth 1 define light_wood <0.6, 0.24, 0.1> define median_wood <0.3, 0.12, 0.03> define dark_wood <0.05, 0.01, 0.005>     define wooden texture { noise surface { ambient 0.2  diffuse 0.7  specular white, 0.5 microfacet Reitz 10 position_fn position_cylindrical position_scale 1  lookup_fn lookup_sawtooth octaves 1 turbulence 1 color_map( [0.0, 0.2, light_wood, light_wood] [0.2, 0.3, light_wood, median_wood] [0.3, 0.4, median_wood, light_wood] [0.4, 0.7, light_wood, light_wood] [0.7, 0.8, light_wood, median_wood] [0.8, 0.9, median_wood, light_wood] [0.9, 1.0, light_wood, dark_wood]) } } define glass texture { surface { ambient 0 diffuse 0 specular 0.2 reflection white, 0.1 transmission white, 1, 1.5 }} define shiny surface { ambient 0.1 diffuse 0.6 specular white, 0.6 microfacet Phong 7  } define steely_blue texture { shiny { color black } } define chrome texture { surface { color white ambient 0.0 diffuse 0.2 specular 0.4 microfacet Phong 10 reflection 0.8 } }   viewpoint {     from <4.000, -1.000, 1.000> at <0.000, 0.000, 0.000> up <0, 1, 0> angle 60     resolution 640, 480 aspect 1.6 image_format 0 }       light <-10, 30, 20> light <-10, 30, -20>   object { disc <0, -2, 0>, <0, 1, 0>, 30 wooden }   object { sphere <0.000, 0.000, 0.000>, 1.00 chrome } object { cylinder <0.000, 0.000, 0.000>, <0.000, 0.000, -4.000>, 0.50 chrome }   After setting up the background and defining colors and textures, the viewpoint is specified. The “camera” is located at a point in 3D space, and it looks towards another point. The angle, image resolution, and aspect ratio are specified. Two lights are present in the image at defined coordinates. The three objects in the image are a wooden disc to represent a table top, and a sphere and cylinder that intersect to form a pin that will be used for the pin board toy in the final animation. When the image is rendered, the following image is produced. The pins are modeled with a chrome surface, so they reflect the environment around them. Note that the scale of the pin shaft is not correct, this will be fixed later. Modeling the Pin Board The frame of the pin-board is made up of three boxes, and six cylinders, the front box is modeled using a clear, slightly reflective solid, with the same refractive index of glass. The other shapes are modeled as metal. object { box <-5.5, -1.5, 1>, <5.5, 5.5, 1.2> glass } object { box <-5.5, -1.5, -0.04>, <5.5, 5.5, -0.09> steely_blue } object { box <-5.5, -1.5, -0.52>, <5.5, 5.5, -0.59> steely_blue } object { cylinder <-5.2, -1.2, 1.4>, <-5.2, -1.2, -0.74>, 0.2 steely_blue } object { cylinder <5.2, -1.2, 1.4>, <5.2, -1.2, -0.74>, 0.2 steely_blue } object { cylinder <-5.2, 5.2, 1.4>, <-5.2, 5.2, -0.74>, 0.2 steely_blue } object { cylinder <5.2, 5.2, 1.4>, <5.2, 5.2, -0.74>, 0.2 steely_blue } object { cylinder <0, -1.2, 1.4>, <0, -1.2, -0.74>, 0.2 steely_blue } object { cylinder <0, 5.2, 1.4>, <0, 5.2, -0.74>, 0.2 steely_blue }   In order to create the matrix of pins that make up the pin board I used a basic console application with a few nested loops to create two intersecting matrixes of pins, which models the layout used in the pin boards. The resulting image is shown below. The pin board contains 11,481 pins, with the scene file containing 23,709 lines of code. For the complete animation 2,000 scene files will be created, which is over 47 million lines of code. Each pin in the pin-board will slide out a specific distance when an object is pressed into the back of the board. This is easily modeled by setting the Z coordinate of the pin to a specific value. In order to set all of the pins in the pin-board to the correct position, a bitmap image can be used. The position of the pin can be set based on the color of the pixel at the appropriate position in the image. When the Windows Azure logo is used to set the Z coordinate of the pins, the following image is generated. The challenge now was to make a cool animation. The Azure Logo is fine, but it is static. Using a normal video to animate the pins would not work; the colors in the video would not be the same as the depth of the objects from the camera. In order to simulate the pin board accurately a series of frames from a depth camera could be used. Windows Kinect The Kenect controllers for the X-Box 360 and Windows feature a depth camera. The Kinect SDK for Windows provides a programming interface for Kenect, providing easy access for .NET developers to the Kinect sensors. The Kinect Explorer provided with the Kinect SDK is a great starting point for exploring Kinect from a developers perspective. Both the X-Box 360 Kinect and the Windows Kinect will work with the Kinect SDK, the Windows Kinect is required for commercial applications, but the X-Box Kinect can be used for hobby projects. The Windows Kinect has the advantage of providing a mode to allow depth capture with objects closer to the camera, which makes for a more accurate depth image for setting the pin positions. Creating a Depth Field Animation The depth field animation used to set the positions of the pin in the pin board was created using a modified version of the Kinect Explorer sample application. In order to simulate the pin board accurately, a small section of the depth range from the depth sensor will be used. Any part of the object in front of the depth range will result in a white pixel; anything behind the depth range will be black. Within the depth range the pixels in the image will be set to RGB values from 0,0,0 to 255,255,255. A screen shot of the modified Kinect Explorer application is shown below. The Kinect Explorer sample application was modified to include slider controls that are used to set the depth range that forms the image from the depth stream. This allows the fine tuning of the depth image that is required for simulating the position of the pins in the pin board. The Kinect Explorer was also modified to record a series of images from the depth camera and save them as a sequence JPEG files that will be used to animate the pins in the animation the Start and Stop buttons are used to start and stop the image recording. En example of one of the depth images is shown below. Once a series of 2,000 depth images has been captured, the task of creating the animation can begin. Rendering a Test Frame In order to test the creation of frames and get an approximation of the time required to render each frame a test frame was rendered on-premise using PolyRay. The output of the rendering process is shown below. The test frame contained 23,629 primitive shapes, most of which are the spheres and cylinders that are used for the 11,800 or so pins in the pin board. The 1280x720 image contains 921,600 pixels, but as anti-aliasing was used the number of rays that were calculated was 4,235,777, with 3,478,754,073 object boundaries checked. The test frame of the pin board with the depth field image applied is shown below. The tracing time for the test frame was 4 minutes 27 seconds, which means rendering the2,000 frames in the animation would take over 148 hours, or a little over 6 days. Although this is much faster that an old 486, waiting almost a week to see the results of an animation would make it challenging for animators to create, view, and refine their animations. It would be much better if the animation could be rendered in less than one hour. Windows Azure Worker Roles The cost of creating an on-premise render farm to render animations increases in proportion to the number of servers. The table below shows the cost of servers for creating a render farm, assuming a cost of $500 per server. Number of Servers Cost 1 $500 16 $8,000 256 $128,000   As well as the cost of the servers, there would be additional costs for networking, racks etc. Hosting an environment of 256 servers on-premise would require a server room with cooling, and some pretty hefty power cabling. The Windows Azure compute services provide worker roles, which are ideal for performing processor intensive compute tasks. With the scalability available in Windows Azure a job that takes 256 hours to complete could be perfumed using different numbers of worker roles. The time and cost of using 1, 16 or 256 worker roles is shown below. Number of Worker Roles Render Time Cost 1 256 hours $30.72 16 16 hours $30.72 256 1 hour $30.72   Using worker roles in Windows Azure provides the same cost for the 256 hour job, irrespective of the number of worker roles used. Provided the compute task can be broken down into many small units, and the worker role compute power can be used effectively, it makes sense to scale the application so that the task is completed quickly, making the results available in a timely fashion. The task of rendering 2,000 frames in an animation is one that can easily be broken down into 2,000 individual pieces, which can be performed by a number of worker roles. Creating a Render Farm in Windows Azure The architecture of the render farm is shown in the following diagram. The render farm is a hybrid application with the following components: ·         On-Premise o   Windows Kinect – Used combined with the Kinect Explorer to create a stream of depth images. o   Animation Creator – This application uses the depth images from the Kinect sensor to create scene description files for PolyRay. These files are then uploaded to the jobs blob container, and job messages added to the jobs queue. o   Process Monitor – This application queries the role instance lifecycle table and displays statistics about the render farm environment and render process. o   Image Downloader – This application polls the image queue and downloads the rendered animation files once they are complete. ·         Windows Azure o   Azure Storage – Queues and blobs are used for the scene description files and completed frames. A table is used to store the statistics about the rendering environment.   The architecture of each worker role is shown below.   The worker role is configured to use local storage, which provides file storage on the worker role instance that can be use by the applications to render the image and transform the format of the image. The service definition for the worker role with the local storage configuration highlighted is shown below. <?xml version="1.0" encoding="utf-8"?> <ServiceDefinition name="CloudRay" >   <WorkerRole name="CloudRayWorkerRole" vmsize="Small">     <Imports>     </Imports>     <ConfigurationSettings>       <Setting name="DataConnectionString" />     </ConfigurationSettings>     <LocalResources>       <LocalStorage name="RayFolder" cleanOnRoleRecycle="true" />     </LocalResources>   </WorkerRole> </ServiceDefinition>     The two executable programs, PolyRay.exe and DTA.exe are included in the Azure project, with Copy Always set as the property. PolyRay will take the scene description file and render it to a Truevision TGA file. As the TGA format has not seen much use since the mid 90’s it is converted to a JPG image using Dave's Targa Animator, another shareware application from the 90’s. Each worker roll will use the following process to render the animation frames. 1.       The worker process polls the job queue, if a job is available the scene description file is downloaded from blob storage to local storage. 2.       PolyRay.exe is started in a process with the appropriate command line arguments to render the image as a TGA file. 3.       DTA.exe is started in a process with the appropriate command line arguments convert the TGA file to a JPG file. 4.       The JPG file is uploaded from local storage to the images blob container. 5.       A message is placed on the images queue to indicate a new image is available for download. 6.       The job message is deleted from the job queue. 7.       The role instance lifecycle table is updated with statistics on the number of frames rendered by the worker role instance, and the CPU time used. The code for this is shown below. public override void Run() {     // Set environment variables     string polyRayPath = Path.Combine(Environment.GetEnvironmentVariable("RoleRoot"), PolyRayLocation);     string dtaPath = Path.Combine(Environment.GetEnvironmentVariable("RoleRoot"), DTALocation);       LocalResource rayStorage = RoleEnvironment.GetLocalResource("RayFolder");     string localStorageRootPath = rayStorage.RootPath;       JobQueue jobQueue = new JobQueue("renderjobs");     JobQueue downloadQueue = new JobQueue("renderimagedownloadjobs");     CloudRayBlob sceneBlob = new CloudRayBlob("scenes");     CloudRayBlob imageBlob = new CloudRayBlob("images");     RoleLifecycleDataSource roleLifecycleDataSource = new RoleLifecycleDataSource();       Frames = 0;       while (true)     {         // Get the render job from the queue         CloudQueueMessage jobMsg = jobQueue.Get();           if (jobMsg != null)         {             // Get the file details             string sceneFile = jobMsg.AsString;             string tgaFile = sceneFile.Replace(".pi", ".tga");             string jpgFile = sceneFile.Replace(".pi", ".jpg");               string sceneFilePath = Path.Combine(localStorageRootPath, sceneFile);             string tgaFilePath = Path.Combine(localStorageRootPath, tgaFile);             string jpgFilePath = Path.Combine(localStorageRootPath, jpgFile);               // Copy the scene file to local storage             sceneBlob.DownloadFile(sceneFilePath);               // Run the ray tracer.             string polyrayArguments =                 string.Format("\"{0}\" -o \"{1}\" -a 2", sceneFilePath, tgaFilePath);             Process polyRayProcess = new Process();             polyRayProcess.StartInfo.FileName =                 Path.Combine(Environment.GetEnvironmentVariable("RoleRoot"), polyRayPath);             polyRayProcess.StartInfo.Arguments = polyrayArguments;             polyRayProcess.Start();             polyRayProcess.WaitForExit();               // Convert the image             string dtaArguments =                 string.Format(" {0} /FJ /P{1}", tgaFilePath, Path.GetDirectoryName (jpgFilePath));             Process dtaProcess = new Process();             dtaProcess.StartInfo.FileName =                 Path.Combine(Environment.GetEnvironmentVariable("RoleRoot"), dtaPath);             dtaProcess.StartInfo.Arguments = dtaArguments;             dtaProcess.Start();             dtaProcess.WaitForExit();               // Upload the image to blob storage             imageBlob.UploadFile(jpgFilePath);               // Add a download job.             downloadQueue.Add(jpgFile);               // Delete the render job message             jobQueue.Delete(jobMsg);               Frames++;         }         else         {             Thread.Sleep(1000);         }           // Log the worker role activity.         roleLifecycleDataSource.Alive             ("CloudRayWorker", RoleLifecycleDataSource.RoleLifecycleId, Frames);     } }     Monitoring Worker Role Instance Lifecycle In order to get more accurate statistics about the lifecycle of the worker role instances used to render the animation data was tracked in an Azure storage table. The following class was used to track the worker role lifecycles in Azure storage.   public class RoleLifecycle : TableServiceEntity {     public string ServerName { get; set; }     public string Status { get; set; }     public DateTime StartTime { get; set; }     public DateTime EndTime { get; set; }     public long SecondsRunning { get; set; }     public DateTime LastActiveTime { get; set; }     public int Frames { get; set; }     public string Comment { get; set; }       public RoleLifecycle()     {     }       public RoleLifecycle(string roleName)     {         PartitionKey = roleName;         RowKey = Utils.GetAscendingRowKey();         Status = "Started";         StartTime = DateTime.UtcNow;         LastActiveTime = StartTime;         EndTime = StartTime;         SecondsRunning = 0;         Frames = 0;     } }     A new instance of this class is created and added to the storage table when the role starts. It is then updated each time the worker renders a frame to record the total number of frames rendered and the total processing time. These statistics are used be the monitoring application to determine the effectiveness of use of resources in the render farm. Rendering the Animation The Azure solution was deployed to Windows Azure with the service configuration set to 16 worker role instances. This allows for the application to be tested in the cloud environment, and the performance of the application determined. When I demo the application at conferences and user groups I often start with 16 instances, and then scale up the application to the full 256 instances. The configuration to run 16 instances is shown below. <?xml version="1.0" encoding="utf-8"?> <ServiceConfiguration serviceName="CloudRay" xmlns="http://schemas.microsoft.com/ServiceHosting/2008/10/ServiceConfiguration" osFamily="1" osVersion="*">   <Role name="CloudRayWorkerRole">     <Instances count="16" />     <ConfigurationSettings>       <Setting name="DataConnectionString"         value="DefaultEndpointsProtocol=https;AccountName=cloudraydata;AccountKey=..." />     </ConfigurationSettings>   </Role> </ServiceConfiguration>     About six minutes after deploying the application the first worker roles become active and start to render the first frames of the animation. The CloudRay Monitor application displays an icon for each worker role instance, with a number indicating the number of frames that the worker role has rendered. The statistics on the left show the number of active worker roles and statistics about the render process. The render time is the time since the first worker role became active; the CPU time is the total amount of processing time used by all worker role instances to render the frames.   Five minutes after the first worker role became active the last of the 16 worker roles activated. By this time the first seven worker roles had each rendered one frame of the animation.   With 16 worker roles u and running it can be seen that one hour and 45 minutes CPU time has been used to render 32 frames with a render time of just under 10 minutes.     At this rate it would take over 10 hours to render the 2,000 frames of the full animation. In order to complete the animation in under an hour more processing power will be required. Scaling the render farm from 16 instances to 256 instances is easy using the new management portal. The slider is set to 256 instances, and the configuration saved. We do not need to re-deploy the application, and the 16 instances that are up and running will not be affected. Alternatively, the configuration file for the Azure service could be modified to specify 256 instances.   <?xml version="1.0" encoding="utf-8"?> <ServiceConfiguration serviceName="CloudRay" xmlns="http://schemas.microsoft.com/ServiceHosting/2008/10/ServiceConfiguration" osFamily="1" osVersion="*">   <Role name="CloudRayWorkerRole">     <Instances count="256" />     <ConfigurationSettings>       <Setting name="DataConnectionString"         value="DefaultEndpointsProtocol=https;AccountName=cloudraydata;AccountKey=..." />     </ConfigurationSettings>   </Role> </ServiceConfiguration>     Six minutes after the new configuration has been applied 75 new worker roles have activated and are processing their first frames.   Five minutes later the full configuration of 256 worker roles is up and running. We can see that the average rate of frame rendering has increased from 3 to 12 frames per minute, and that over 17 hours of CPU time has been utilized in 23 minutes. In this test the time to provision 140 worker roles was about 11 minutes, which works out at about one every five seconds.   We are now half way through the rendering, with 1,000 frames complete. This has utilized just under three days of CPU time in a little over 35 minutes.   The animation is now complete, with 2,000 frames rendered in a little over 52 minutes. The CPU time used by the 256 worker roles is 6 days, 7 hours and 22 minutes with an average frame rate of 38 frames per minute. The rendering of the last 1,000 frames took 16 minutes 27 seconds, which works out at a rendering rate of 60 frames per minute. The frame counts in the server instances indicate that the use of a queue to distribute the workload has been very effective in distributing the load across the 256 worker role instances. The first 16 instances that were deployed first have rendered between 11 and 13 frames each, whilst the 240 instances that were added when the application was scaled have rendered between 6 and 9 frames each.   Completed Animation I’ve uploaded the completed animation to YouTube, a low resolution preview is shown below. Pin Board Animation Created using Windows Kinect and 256 Windows Azure Worker Roles   The animation can be viewed in 1280x720 resolution at the following link: http://www.youtube.com/watch?v=n5jy6bvSxWc Effective Use of Resources According to the CloudRay monitor statistics the animation took 6 days, 7 hours and 22 minutes CPU to render, this works out at 152 hours of compute time, rounded up to the nearest hour. As the usage for the worker role instances are billed for the full hour, it may have been possible to render the animation using fewer than 256 worker roles. When deciding the optimal usage of resources, the time required to provision and start the worker roles must also be considered. In the demo I started with 16 worker roles, and then scaled the application to 256 worker roles. It would have been more optimal to start the application with maybe 200 worker roles, and utilized the full hour that I was being billed for. This would, however, have prevented showing the ease of scalability of the application. The new management portal displays the CPU usage across the worker roles in the deployment. The average CPU usage across all instances is 93.27%, with over 99% used when all the instances are up and running. This shows that the worker role resources are being used very effectively. Grid Computing Scenarios Although I am using this scenario for a hobby project, there are many scenarios where a large amount of compute power is required for a short period of time. Windows Azure provides a great platform for developing these types of grid computing applications, and can work out very cost effective. ·         Windows Azure can provide massive compute power, on demand, in a matter of minutes. ·         The use of queues to manage the load balancing of jobs between role instances is a simple and effective solution. ·         Using a cloud-computing platform like Windows Azure allows proof-of-concept scenarios to be tested and evaluated on a very low budget. ·         No charges for inbound data transfer makes the uploading of large data sets to Windows Azure Storage services cost effective. (Transaction charges still apply.) Tips for using Windows Azure for Grid Computing Scenarios I found the implementation of a render farm using Windows Azure a fairly simple scenario to implement. I was impressed by ease of scalability that Azure provides, and by the short time that the application took to scale from 16 to 256 worker role instances. In this case it was around 13 minutes, in other tests it took between 10 and 20 minutes. The following tips may be useful when implementing a grid computing project in Windows Azure. ·         Using an Azure Storage queue to load-balance the units of work across multiple worker roles is simple and very effective. The design I have used in this scenario could easily scale to many thousands of worker role instances. ·         Windows Azure accounts are typically limited to 20 cores. If you need to use more than this, a call to support and a credit card check will be required. ·         Be aware of how the billing model works. You will be charged for worker role instances for the full clock our in which the instance is deployed. Schedule the workload to start just after the clock hour has started. ·         Monitor the utilization of the resources you are provisioning, ensure that you are not paying for worker roles that are idle. ·         If you are deploying third party applications to worker roles, you may well run into licensing issues. Purchasing software licenses on a per-processor basis when using hundreds of processors for a short time period would not be cost effective. ·         Third party software may also require installation onto the worker roles, which can be accomplished using start-up tasks. Bear in mind that adding a startup task and possible re-boot will add to the time required for the worker role instance to start and activate. An alternative may be to use a prepared VM and use VM roles. ·         Consider using the Windows Azure Autoscaling Application Block (WASABi) to autoscale the worker roles in your application. When using a large number of worker roles, the utilization must be carefully monitored, if the scaling algorithms are not optimal it could get very expensive!

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  • Making a Statement: How to retrieve the T-SQL statement that caused an event

    - by extended_events
    If you’ve done any troubleshooting of T-SQL, you know that sooner or later, probably sooner, you’re going to want to take a look at the actual statements you’re dealing with. In extended events we offer an action (See the BOL topic that covers Extended Events Objects for a description of actions) named sql_text that seems like it is just the ticket. Well…not always – sounds like a good reason for a blog post. When is a statement not THE statement? The sql_text action returns the same information that is returned from DBCC INPUTBUFFER, which may or may not be what you want. For example, if you execute a stored procedure, the sql_text action will return something along the lines of “EXEC sp_notwhatiwanted” assuming that is the statement you sent from the client. Often times folks would like something more specific, like the actual statements that are being run from within the stored procedure or batch. Enter the stack Extended events offers another action, this one with the descriptive name of tsql_stack, that includes the sql_handle and offset information about the statements being run when an event occurs. With the sql_handle and offset values you can retrieve the specific statement you seek using the DMV dm_exec_sql_statement. The BOL topic for dm_exec_sql_statement provides an example for how to extract this information, so I’ll cover the gymnastics required to get the sql_handle and offset values out of the tsql_stack data collected by the action. I’m the first to admit that this isn’t pretty, but this is what we have in SQL Server 2008 and 2008 R2. We will be making it easier to get statement level information in the next major release of SQL Server. The sample code For this example I have a stored procedure that includes multiple statements and I have a need to differentiate between those two statements in my tracing. I’m going to track two events: module_end tracks the completion of the stored procedure execution and sp_statement_completed tracks the execution of each statement within a stored procedure. I’m adding the tsql_stack events (since that’s the topic of this post) and the sql_text action for comparison sake. (If you have questions about creating event sessions, check out Pedro’s post Introduction to Extended Events.) USE AdventureWorks2008GO -- Test SPCREATE PROCEDURE sp_multiple_statementsASSELECT 'This is the first statement'SELECT 'this is the second statement'GO -- Create a session to look at the spCREATE EVENT SESSION track_sprocs ON SERVERADD EVENT sqlserver.module_end (ACTION (sqlserver.tsql_stack, sqlserver.sql_text)),ADD EVENT sqlserver.sp_statement_completed (ACTION (sqlserver.tsql_stack, sqlserver.sql_text))ADD TARGET package0.ring_bufferWITH (MAX_DISPATCH_LATENCY = 1 SECONDS)GO -- Start the sessionALTER EVENT SESSION track_sprocs ON SERVERSTATE = STARTGO -- Run the test procedureEXEC sp_multiple_statementsGO -- Stop collection of events but maintain ring bufferALTER EVENT SESSION track_sprocs ON SERVERDROP EVENT sqlserver.module_end,DROP EVENT sqlserver.sp_statement_completedGO Aside: Altering the session to drop the events is a neat little trick that allows me to stop collection of events while keeping in-memory targets such as the ring buffer available for use. If you stop the session the in-memory target data is lost. Now that we’ve collected some events related to running the stored procedure, we need to do some processing of the data. I’m going to do this in multiple steps using temporary tables so you can see what’s going on; kind of like having to “show your work” on a math test. The first step is to just cast the target data into XML so I can work with it. After that you can pull out the interesting columns, for our purposes I’m going to limit the output to just the event name, object name, stack and sql text. You can see that I’ve don a second CAST, this time of the tsql_stack column, so that I can further process this data. -- Store the XML data to a temp tableSELECT CAST( t.target_data AS XML) xml_dataINTO #xml_event_dataFROM sys.dm_xe_sessions s INNER JOIN sys.dm_xe_session_targets t    ON s.address = t.event_session_addressWHERE s.name = 'track_sprocs' SELECT * FROM #xml_event_data -- Parse the column data out of the XML blockSELECT    event_xml.value('(./@name)', 'varchar(100)') as [event_name],    event_xml.value('(./data[@name="object_name"]/value)[1]', 'varchar(255)') as [object_name],    CAST(event_xml.value('(./action[@name="tsql_stack"]/value)[1]','varchar(MAX)') as XML) as [stack_xml],    event_xml.value('(./action[@name="sql_text"]/value)[1]', 'varchar(max)') as [sql_text]INTO #event_dataFROM #xml_event_data    CROSS APPLY xml_data.nodes('//event') n (event_xml) SELECT * FROM #event_data event_name object_name stack_xml sql_text sp_statement_completed NULL <frame level="1" handle="0x03000500D0057C1403B79600669D00000100000000000000" line="4" offsetStart="94" offsetEnd="172" /><frame level="2" handle="0x01000500CF3F0331B05EC084000000000000000000000000" line="1" offsetStart="0" offsetEnd="-1" /> EXEC sp_multiple_statements sp_statement_completed NULL <frame level="1" handle="0x03000500D0057C1403B79600669D00000100000000000000" line="6" offsetStart="174" offsetEnd="-1" /><frame level="2" handle="0x01000500CF3F0331B05EC084000000000000000000000000" line="1" offsetStart="0" offsetEnd="-1" /> EXEC sp_multiple_statements module_end sp_multiple_statements <frame level="1" handle="0x03000500D0057C1403B79600669D00000100000000000000" line="0" offsetStart="0" offsetEnd="0" /><frame level="2" handle="0x01000500CF3F0331B05EC084000000000000000000000000" line="1" offsetStart="0" offsetEnd="-1" /> EXEC sp_multiple_statements After parsing the columns it’s easier to see what is recorded. You can see that I got back two sp_statement_completed events, which makes sense given the test procedure I’m running, and I got back a single module_end for the entire statement. As described, the sql_text isn’t telling me what I really want to know for the first two events so a little extra effort is required. -- Parse the tsql stack information into columnsSELECT    event_name,    object_name,    frame_xml.value('(./@level)', 'int') as [frame_level],    frame_xml.value('(./@handle)', 'varchar(MAX)') as [sql_handle],    frame_xml.value('(./@offsetStart)', 'int') as [offset_start],    frame_xml.value('(./@offsetEnd)', 'int') as [offset_end]INTO #stack_data    FROM #event_data        CROSS APPLY    stack_xml.nodes('//frame') n (frame_xml)    SELECT * from #stack_data event_name object_name frame_level sql_handle offset_start offset_end sp_statement_completed NULL 1 0x03000500D0057C1403B79600669D00000100000000000000 94 172 sp_statement_completed NULL 2 0x01000500CF3F0331B05EC084000000000000000000000000 0 -1 sp_statement_completed NULL 1 0x03000500D0057C1403B79600669D00000100000000000000 174 -1 sp_statement_completed NULL 2 0x01000500CF3F0331B05EC084000000000000000000000000 0 -1 module_end sp_multiple_statements 1 0x03000500D0057C1403B79600669D00000100000000000000 0 0 module_end sp_multiple_statements 2 0x01000500CF3F0331B05EC084000000000000000000000000 0 -1 Parsing out the stack information doubles the fun and I get two rows for each event. If you examine the stack from the previous table, you can see that each stack has two frames and my query is parsing each event into frames, so this is expected. There is nothing magic about the two frames, that’s just how many I get for this example, it could be fewer or more depending on your statements. The key point here is that I now have a sql_handle and the offset values for those handles, so I can use dm_exec_sql_statement to get the actual statement. Just a reminder, this DMV can only return what is in the cache – if you have old data it’s possible your statements have been ejected from the cache. “Old” is a relative term when talking about caches and can be impacted by server load and how often your statement is actually used. As with most things in life, your mileage may vary. SELECT    qs.*,     SUBSTRING(st.text, (qs.offset_start/2)+1,         ((CASE qs.offset_end          WHEN -1 THEN DATALENGTH(st.text)         ELSE qs.offset_end         END - qs.offset_start)/2) + 1) AS statement_textFROM #stack_data AS qsCROSS APPLY sys.dm_exec_sql_text(CONVERT(varbinary(max),sql_handle,1)) AS st event_name object_name frame_level sql_handle offset_start offset_end statement_text sp_statement_completed NULL 1 0x03000500D0057C1403B79600669D00000100000000000000 94 172 SELECT 'This is the first statement' sp_statement_completed NULL 1 0x03000500D0057C1403B79600669D00000100000000000000 174 -1 SELECT 'this is the second statement' module_end sp_multiple_statements 1 0x03000500D0057C1403B79600669D00000100000000000000 0 0 C Now that looks more like what we were after, the statement_text field is showing the actual statement being run when the sp_statement_completed event occurs. You’ll notice that it’s back down to one row per event, what happened to frame 2? The short answer is, “I don’t know.” In SQL Server 2008 nothing is returned from dm_exec_sql_statement for the second frame and I believe this to be a bug; this behavior has changed in the next major release and I see the actual statement run from the client in frame 2. (In other words I see the same statement that is returned by the sql_text action  or DBCC INPUTBUFFER) There is also something odd going on with frame 1 returned from the module_end event; you can see that the offset values are both 0 and only the first letter of the statement is returned. It seems like the offset_end should actually be –1 in this case and I’m not sure why it’s not returning this correctly. This behavior is being investigated and will hopefully be corrected in the next major version. You can workaround this final oddity by ignoring the offsets and just returning the entire cached statement. SELECT    event_name,    sql_handle,    ts.textFROM #stack_data    CROSS APPLY sys.dm_exec_sql_text(CONVERT(varbinary(max),sql_handle,1)) as ts event_name sql_handle text sp_statement_completed 0x0300070025999F11776BAF006F9D00000100000000000000 CREATE PROCEDURE sp_multiple_statements AS SELECT 'This is the first statement' SELECT 'this is the second statement' sp_statement_completed 0x0300070025999F11776BAF006F9D00000100000000000000 CREATE PROCEDURE sp_multiple_statements AS SELECT 'This is the first statement' SELECT 'this is the second statement' module_end 0x0300070025999F11776BAF006F9D00000100000000000000 CREATE PROCEDURE sp_multiple_statements AS SELECT 'This is the first statement' SELECT 'this is the second statement' Obviously this gives more than you want for the sp_statement_completed events, but it’s the right information for module_end. I leave it to you to determine when this information is needed and use the workaround when appropriate. Aside: You might think it’s odd that I’m showing apparent bugs with my samples, but you’re going to see this behavior if you use this method, so you need to know about it.I’m all about transparency. Happy Eventing- Mike Share this post: email it! | bookmark it! | digg it! | reddit! | kick it! | live it!

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  • The UIManager Pattern

    - by Duncan Mills
    One of the most common mistakes that I see when reviewing ADF application code, is the sin of storing UI component references, most commonly things like table or tree components in Session or PageFlow scope. The reasons why this is bad are simple; firstly, these UI object references are not serializable so would not survive a session migration between servers and secondly there is no guarantee that the framework will re-use the same component tree from request to request, although in practice it generally does do so. So there danger here is, that at best you end up with an NPE after you session has migrated, and at worse, you end up pinning old generations of the component tree happily eating up your precious memory. So that's clear, we should never. ever, be storing references to components anywhere other than request scope (or maybe backing bean scope). So double check the scope of those binding attributes that map component references into a managed bean in your applications.  Why is it Such a Common Mistake?  At this point I want to examine why there is this urge to hold onto these references anyway? After all, JSF will obligingly populate your backing beans with the fresh and correct reference when needed.   In most cases, it seems that the rational is down to a lack of distinction within the application between what is data and what is presentation. I think perhaps, a cause of this is the logical separation between business data behind the ADF data binding (#{bindings}) façade and the UI components themselves. Developers tend to think, OK this is my data layer behind the bindings object and everything else is just UI.  Of course that's not the case.  The UI layer itself will have state which is intrinsically linked to the UI presentation rather than the business model, but at the same time should not be tighly bound to a specific instance of any single UI component. So here's the problem.  I think developers try and use the UI components as state-holders for this kind of data, rather than using them to represent that state. An example of this might be something like the selection state of a tabset (panelTabbed), you might be interested in knowing what the currently disclosed tab is. The temptation that leads to the component reference sin is to go and ask the tabset what the selection is.  That of course is fine in context - e.g. a handler within the same request scoped bean that's got the binding to the tabset. However, it leads to problems when you subsequently want the same information outside of the immediate scope.  The simple solution seems to be to chuck that component reference into session scope and then you can simply re-check in the same way, leading of course to this mistake. Turn it on its Head  So the correct solution to this is to turn the problem on its head. If you are going to be interested in the value or state of some component outside of the immediate request context then it becomes persistent state (persistent in the sense that it extends beyond the lifespan of a single request). So you need to externalize that state outside of the component and have the component reference and manipulate that state as needed rather than owning it. This is what I call the UIManager pattern.  Defining the Pattern The  UIManager pattern really is very simple. The premise is that every application should define a session scoped managed bean, appropriately named UIManger, which is specifically responsible for holding this persistent UI component related state.  The actual makeup of the UIManger class varies depending on a needs of the application and the amount of state that needs to be stored. Generally I'll start off with a Map in which individual flags can be created as required, although you could opt for a more formal set of typed member variables with getters and setters, or indeed a mix. This UIManager class is defined as a session scoped managed bean (#{uiManager}) in the faces-config.xml.  The pattern is to then inject this instance of the class into any other managed bean (usually request scope) that needs it using a managed property.  So typically you'll have something like this:   <managed-bean>     <managed-bean-name>uiManager</managed-bean-name>     <managed-bean-class>oracle.demo.view.state.UIManager</managed-bean-class>     <managed-bean-scope>session</managed-bean-scope>   </managed-bean>  When is then injected into any backing bean that needs it:    <managed-bean>     <managed-bean-name>mainPageBB</managed-bean-name>     <managed-bean-class>oracle.demo.view.MainBacking</managed-bean-class>     <managed-bean-scope>request</managed-bean-scope>     <managed-property>       <property-name>uiManager</property-name>       <property-class>oracle.demo.view.state.UIManager</property-class>       <value>#{uiManager}</value>     </managed-property>   </managed-bean> In this case the backing bean in question needs a member variable to hold and reference the UIManager: private UIManager _uiManager;  Which should be exposed via a getter and setter pair with names that match the managed property name (e.g. setUiManager(UIManager _uiManager), getUiManager()).  This will then give your code within the backing bean full access to the UI state. UI components in the page can, of course, directly reference the uiManager bean in their properties, for example, going back to the tab-set example you might have something like this: <af:paneltabbed>   <af:showDetailItem text="First"                disclosed="#{uiManager.settings['MAIN_TABSET_STATE'].['FIRST']}"> ...   </af:showDetailItem>   <af:showDetailItem text="Second"                      disclosed="#{uiManager.settings['MAIN_TABSET_STATE'].['SECOND']}">     ...   </af:showDetailItem>   ... </af:panelTabbed> Where in this case the settings member within the UI Manger is a Map which contains a Map of Booleans for each tab under the MAIN_TABSET_STATE key. (Just an example you could choose to store just an identifier for the selected tab or whatever, how you choose to store the state within UI Manger is up to you.) Get into the Habit So we can see that the UIManager pattern is not great strain to implement for an application and can even be retrofitted to an existing application with ease. The point is, however, that you should always take this approach rather than committing the sin of persistent component references which will bite you in the future or shotgun scattered UI flags on the session which are hard to maintain.  If you take the approach of always accessing all UI state via the uiManager, or perhaps a pageScope focused variant of it, you'll find your applications much easier to understand and maintain. Do it today!

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  • Day 6 - Game Menuing Woes and Future Screen Sneak Peeks

    - by dapostolov
    So, after my last post on Day 5 I dabbled with my game class design. I took the approach where each game objects is tightly coupled with a graphic. The good news is I got the menu working but not without some hard knocks and game growing pains. I'll explain later, but for now...here is a class diagram of my first stab at my class structure and some code...   Ok, there are few mistakes, however, I'm going to leave it as is for now... As you can see I created an inital abstract base class called GameSprite. This class when inherited will provide a simple virtual default draw method:        public virtual void DrawSprite(SpriteBatch spriteBatch)         {             spriteBatch.Draw(Sprite, Position, Color.White);         } The benefits of coding it this way allows me to inherit the class and utilise the method in the screen draw method...So regardless of what the graphic object type is it will now have the ability to render a static image on the screen. Example: public class MyStaticTreasureChest : GameSprite {} If you remember the window draw method from Day 3's post, we could use the above code as follows...         protected override void Draw(GameTime gameTime)         {             GraphicsDevice.Clear(Color.CornflowerBlue);             spriteBatch.Begin(SpriteBlendMode.AlphaBlend);             foreach(var gameSprite in ListOfGameObjects)            {                 gameSprite.DrawSprite(spriteBatch);            }             spriteBatch.End();             base.Draw(gameTime);         } I have to admit the GameSprite object is pretty plain as with its DrawSprite method... But ... we now have the ability to render 3 static menu items on the screen ... BORING! I want those menu items to do something exciting, which of course involves animation... So, let's have a peek at AnimatedGameSprite in the above game diagram. The idea with the AnimatedGameSprite is that it has an image to animate...such as ... characters, fireballs, and... menus! So after inheriting from GameSprite class, I added a few more options such as UpdateSprite...         public virtual void UpdateSprite(float elapsed)         {             _totalElapsed += elapsed;             if (_totalElapsed > _timePerFrame)             {                 _frame++;                 _frame = _frame % _framecount;                 _totalElapsed -= _timePerFrame;             }         }  And an overidden DrawSprite...         public override void DrawSprite(SpriteBatch spriteBatch)         {             int FrameWidth = Sprite.Width / _framecount;             Rectangle sourcerect = new Rectangle(FrameWidth * _frame, 0, FrameWidth, Sprite.Height);             spriteBatch.Draw(Sprite, Position, sourcerect, Color.White, _rotation, _origin, _scale, SpriteEffects.None, _depth);         } With these two methods...I can animate and image, all I had to do was add a few more lines to the screens Update Method (From Day 3), like such:             float elapsed = (float) gameTime.ElapsedGameTime.TotalSeconds;             foreach (var item in ListOfAnimatedGameObjects)             {                 item.UpdateSprite(elapsed);             } And voila! My images begin to animate in one spot, on the screen... Hmm, but how do I interact with the menu items using a mouse...well the mouse cursor was easy enough... this.IsMouseVisible = true; But, to have it "interact" with an image was a bit more tricky...I had to perform collision detection!             mouseStateCurrent = Mouse.GetState();             var uiEnabledSprites = (from s in menuItems                                    where s.IsEnabled                                    select s).ToList();             foreach (var item in uiEnabledSprites)             {                 var r = new Rectangle((int)item.Position.X, (int)item.Position.Y, item.Sprite.Width, item.Sprite.Height);                 item.MenuState = MenuState.Normal;                 if (r.Intersects(new Rectangle(mouseStateCurrent.X, mouseStateCurrent.Y, 0, 0)))                 {                     item.MenuState = MenuState.Hover;                     if (mouseStatePrevious.LeftButton == ButtonState.Pressed                         && mouseStateCurrent.LeftButton == ButtonState.Released)                     {                         item.MenuState = MenuState.Pressed;                     }                 }             }             mouseStatePrevious = mouseStateCurrent; So, basically, what it is doing above is iterating through all my interactive objects and detecting a rectangle collision and the object , plays the state animation (or static image).  Lessons Learned, Time Burned... So, I think I did well to start, but after I hammered out my prototype...well...things got sloppy and I began to realise some design flaws... At the time: I couldn't seem to figure out how to open another window, such as the character creation screen Input was not event based and it was bugging me My menu design relied heavily on mouse input and I couldn't use keyboard. Mouse input, is tightly bound with graphic rendering / positioning, so its logic will have to be in each scene. Menu animations would stop mid frame, then continue when the action occured again. This is bad, because...what if I had a sword sliding onthe screen? Then it would slide a quarter of the way, then stop due to another action, then render again mid-slide... it just looked sloppy. Menu, Solved!? To solve the above problems I did a little research and I found some great code in the XNA forums. The one worth mentioning was the GameStateManagementSample. With this sample, you can create a basic "text based" menu system which allows you to swap screens, popup screens, play the game, and quit....basic game state management... In my next post I'm going to dwelve a bit more into this code and adapt it with my code from this prototype. Text based menus just won't cut it for me, for now...however, I'm still going to stick with my animated menu item idea. A sneak peek using the Game State Management Sample...with no changes made... Cool Things to Mention: At work ... I tend to break out in random conversations every-so-often and I get talking about some of my challenges with this game (or some stupid observation about something... stupid) During one conversation I was discussing how I should animate my images; I explained that I knew I had to use the Update method provided, but I didn't know how (at the time) to render an image at an appropriate "pace" and how many frames to use, etc.. I also got thinking that if a machine rendered my images faster / slower, that was surely going to f-up my animations. To which a friend, Sheldon,  answered, surely the Draw method is like a camera taking a snapshot of a scene in time. Then it clicked...I understood the big picture of the game engine... After some research I discovered that the Draw method attempts to keep a framerate of 60 fps. From what I understand, the game engine will even leave out a few calls to the draw method if it begins to slow down. This is why we want to put our sprite updates in the update method. Then using a game timer (provided by the engine), we want to render the scene based on real time passed, not framerate. So even the engine renders at 20 fps, the animations will still animate at the same real time speed! Which brings up another point. Why 60 fps? I'm speculating that Microsoft capped it because LCD's dont' refresh faster than 60 fps? On another note, If the game engine knows its falling behind in rendering...then surely we can harness this to speed up our games. Maybe I can find some flag which tell me if the game is lagging, and what the current framerate is, etc...(instead of coding it like I did last time) Sheldon, suggested maybe I can render like WoW does, in prioritised layers...I think he's onto something, however I don't think I'll have that many graphics to worry about such a problem of graphic latency. We'll see. People to Mention: Well,as you are aware I hadn't posted in a couple days and I was surprised to see a few emails and messenger queries about my game progress (and some concern as to why I stopped). I want to thank everyone for their kind words of support and put everyone at ease by stating that I do intend on completing this project. Granted I only have a few hours each night, but, I'll do it. Thank you to Garth for mailing in my next screen! That was a nice surprise! The Sneek Peek you've been waiting for... Garth has also volunteered to render me some wizard images. He was a bit shocked when I asked for them in 2D animated strips. He said I was going backward (and that I have really bad Game Development Lingo). But, I advised Garth that I will use 3D images later...for now...2D images. Garth also had some great game design ideas to add on. I advised him that I will save his ideas and include them in the future design document (for the 3d version?). Lastly, my best friend Alek, is going to join me in developing this game. This was a project we started eons ago but never completed because of our careers. Now, priorities change and we have some spare time on our hands. Let's see what trouble Alek and I can get into! Tonight I'll be uploading my prototypes and base game to a source control for both of us to work off of. D.

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  • Timeout Considerations for Solicit Response

    - by Michael Stephenson
    Background One of the clients I work with had been experiencing some issues for a while surrounding web service timeouts.  It's been a little challenging to work through the problems due to limitations in the diagnostic information available from one of the applications, but I learned some interesting things while troubleshooting the problem which don't seem to have been discussed much in the community so I thought I'd share my findings. In the scenario we have BizTalk trying to make calls to a .net web service which was exposed as a WSE 2 endpoint.  In the process BizTalk will try to make a large number of concurrent web service calls to the application, and the backend application has more than enough infrastructure and capability to handle the load. We have configured the <ConnectionManagement> section of the BizTalk configuration file to support up to 100 concurrent connections from each of our 2 BizTalk send servers to the web servers of the application. The problem we were facing was that the BizTalk side was reporting a significant number of timeouts when calling the web service.   One of the biggest issues was the challenge of being able to correlate a message from BizTalk to the IIS log in the .net application and the custom logs in the application especially when there was a fairly large number of servers hosting the web services.  However the key moment came when we were able to identify a specific call which had taken 40 seconds to execute on the server (yes a long time I know but that's a different story!).  Anyway we were able to identify that this had timed out on the BizTalk side.  Based on the normal 2 minute timeout we knew something unexpected was going on. From here I decided to do some experimentation and I wanted to start outside of BizTalk because my hunch was this was not a BizTalk behaviour but something which was being highlighted by BizTalk because of our large load.     Server-side - Sample Web Service To begin with I created a sample web service.  Nothing special just a vanilla asmx web service hosted in IIS6 on Windows 2003 Standard Edition.  The web service is just a hello world style web service as shown in the below picture.  The only key feature is that the server side web method has a 30 second sleep in it and will trace out some information before and after the thread is set to sleep.      In the configuration for this web service there again is nothing special it's pretty much the most plain simple web service you could build. Client-Side To begin looking at what was happening with our example I created a number of different ways to consume the web service. SoapHttpClientProtocol Example I created a small application which would use a normal proxy generated to call the web service.  It would iterate around a loop and make calls using the begin/end methods so I can do this asynchronously.  I would do a loop of 20 calls with the ConnectionManager configuration section supporting only 5 concurrent connections to the server.     <connectionManagement> <remove address="*"/> <add address = "*" maxconnection = "12" /> <add address = "http://<ServerName>" maxconnection = "5" />                         </connectionManagement> </system.net>     The below picture shows an example of the service calling code, key points are: I have configured the timeout of 40 seconds for the proxy I am using the asynchronous methods on the proxy to call the web service         The Test I would run the client and execute 21 calls to the web service.   The Results  Below is the client side trace showing what's happening on the client. In the below diagram is the web service side trace showing what's happening on the server Some observations on the results are: All of the calls were successful from the clients perspective You could see the next call starting on the server as soon as the previous one had completed Calls took significantly longer than 40 seconds from the start of our call to the return. In fact call 20 took 2 minutes and 30 seconds from the perspective of my code to execute even though I had set the timeout to 40 seconds     WSE 2 Sample In the second example I used the exact same code to call the web service again with a single exception that I modified the web service proxy to derive from WebServiceClient protocol which is part of WSE 2 (using SP3).  The below picture shows the basic code and the key points are: I have configured the timeout of 40 seconds for the proxy I am using the asynchronous methods on the proxy to call the web service        The Test This test would execute 21 calls from the client to the web service.   The Results  The below trace is from the client side: The below trace is from the server side:   Some observations on the trace results for this scenario are: With call 4 if you look at the server side trace it did not start executing on the server for a number of seconds after the other 4 initial calls which were accepted by the server. I re-ran the test and this happened a couple of times and not on most others so at this point I'm just putting this down to something unexpected happening on the development machine and we will leave this observation out of scope of this article. You can see that the client side trace statement executed almost immediately in all cases All calls after the initial few calls would timeout On the client side the calls that did timeout; timed out in a longer duration than the 40 seconds we set as the timeout You can see that as calls were completing on the server the next calls were starting to come through The calls that timed out on the client did actually connect to the server and their server side execution completed successfully     Elaboration on the findings Based on the above observations I have drawn the below sequence diagram to illustrate conceptually what is happening.  Everything except the final web service object is on the client side of the call. In the diagram below I've put two notes on the Web Service Proxy to show the two different places where the different base classes seem to start their timeout counters. From the earlier samples we can work out that the timeout counter for the WSE web service proxy starts before the one for the SoapHttpClientProtocol proxy and the WSE one includes the time to get a connection from the pool; whereas the Soap proxy timeout just covers the method execution. One interesting observation is if we rerun the above sample and increase the number of calls from 21 to 100,000 then for the WSE sample we will see a similar pattern where everything after the first few calls will timeout on the client as soon as it makes a connection to the server whereas the soap proxy will happily plug away and process all of the calls without a single timeout. I have actually set the sample running overnight and this did happen. At this point you are probably thinking the same thoughts I was at the time about the differences in behaviour and which is right and why are they different? I'm not sure there is a definitive answer to this in the documentation, or at least not that I could find! I think you just have to consider that they are different and they could have different effects depending on your messaging solution. In lots of situations this is just not an issue as your concurrent requests doesn't get to the situation where you end up throttling the web service calls on the client side, however this is definitely more common with an integration broker such as BizTalk where you often have high throughput requirements.  Some of the considerations you should make Based on this behaviour you should be aware of the following: In a .net application if you are making lots of concurrent web service calls from an application in an asynchronous manner your user may thing they are experiencing poor performance but you think your web service is working well. The problem could be that the client will have a default of 2 connections to remote servers so you should bear this in mind When you are developing a BizTalk solution or a .net solution with the WSE 2 stack you may experience timeouts under load and throttling the number of connections using the max connections element in the configuration file will not help you For an application using WSE2 or SoapHttpClientProtocol an expired timeout will not throw an error until after a connection to the server has been made so you should consider this in your transaction and durability patterns     Our Work Around In the short term for our specific scenario we know that we can handle this by just increasing our timeout value.  There is only a specific small window when we get lots of concurrent traffic that causes this scenario so we should be able to increase the timeout to take into consideration the additional client side wait, and on the odd occasion where we do get a timeout the BizTalk send port retry will handle this. What was causing our original problem was that for that short window we were getting a lot of retries which significantly increased the load on our send servers and highlighted the issue.  Longer Term Solution As a longer term solution this really gives us more ammunition to argue a migration to WCF. The application we are calling has some factors which limit the protocols we can use but with WCF we would have more control on the various timeout options because in WCF you can configure specific parts of the timeout. Summary I've had this blog post on my to do list for ages but hopefully it will be useful to some people to just understand this behaviour and to possibly help you with some performance issues you may have. I do not believe there is too much in the way of documentation particularly around WSE2 and ASMX in this area so again another bit of ammunition for migrating to WCF. I'll try to do a follow up post with the sample for WCF to show how this changes things.

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  • Navigation in Win8 Metro Style applications

    - by Dennis Vroegop
    In Windows 8, Touch is, as they say, a first class citizen. Now, to be honest: they also said that in Windows 7. However in Win8 this is actually true. Applications are meant to be used by touch. Yes, you can still use mouse, keyboard and pen and your apps should take that into account but touch is where you should focus on initially. Will all users have touch enabled devices? No, not in the first place. I don’t think touchscreens will be on every device sold next year. But in 5 years? Who knows? Don’t forget: if your app is successful it will be around for a long time and by that time touchscreens will be everywhere. Another reason to embrace touch is that it’s easier to develop a touch-oriented app and then to make sure that keyboard, nouse and pen work as doing it the other way around. Porting a mouse-based application to a touch based application almost never works. The reverse gives you much more chances for success. That being said, there are some things that you need to think about. Most people have more than one finger, while most users only use one mouse at the time. Still, most touch-developers translate their mouse-knowledge to the touch and think they did a good job. Martin Tirion from Microsoft said that since Touch is a new language people face the same challenges they do when learning a new real spoken language. The first thing people try when learning a new language is simply replace the words in their native language to the newly learned words. At first they don’t care about grammar. To a native speaker of that other language this sounds all wrong but they still will be able to understand what the intention was. If you don’t believe me: try Google translate to translate something for you from your language to another and then back and see what happens. The same thing happens with Touch. Most developers translate a mouse-click into a tap-event and think they’re done. Well matey, you’re not done. Not by far. There are things you can do with a mouse that you cannot do with touch. Think hover. A mouse has the ability to ‘slide’ over UI elements. Touch doesn’t (I know: with Pen you can do this but I’m talking about actual fingers here). A touch is either there or it isn’t. And right-click? Forget about it. A click is a click.  Yes, you have more than one finger but the machine doesn’t know which finger you use… The other way around is also true. Like I said: most users only have one mouse but they are likely to have more than one finger. So how do we take that into account? Thinking about this is really worth the time: you might come up with some surprisingly good ideas! Still: don’t forget that not every user has touch-enabled hardware so make sure your app is useable for both groups. Keep this in mind: we’re going to need it later on! Now. Apps should be easy to use. You don’t want your user to read through pages and pages of documentation before they can use the app. Imagine that spotter next to an airfield suddenly seeing a prototype of a Concorde 2 landing on the nearby runway. He probably wants to enter that information in our app NOW and not after he’s taken a 3 day course. Even if he still has to download the app, install it for the first time and then run it he should be on his way immediately. At least, fast enough to note down the details of that unique, rare and possibly exciting sighting he just did. So.. How do we do this? Well, I am not talking about games here. Games are in a league of their own. They fall outside the scope of the apps I am describing. But all the others can roughly be characterized as being one of two flavors: the navigation is either flat or hierarchical. That’s it. And if it’s hierarchical it’s no more than three levels deep. Not more. Your users will get lost otherwise and we don’t want that. Flat is simple. Just imagine we have one screen that is as high as our physical screen is and as wide as you need it to be. Don’t worry if it doesn’t fit on the screen: people can scroll to the right and left. Don’t combine up/down and left/right scrolling: it’s confusing. Next to that, since most users will hold their device in landscape mode it’s very natural to scroll horizontal. So let’s use that when we have a flat model. The same applies to the hierarchical model. Try to have at most three levels. If you need more space, find a way to group the items in such a way that you can fit it in three, very wide lanes. At the highest level we have the so called hub level. This is the entry point of the app and as such it should give the user an immediate feeling of what the app is all about. If your app has categories if items then you might show these categories here. And while you’re at it: also show 2 or 3 of the items itself here to give the user a taste of what lies beneath. If the user selects a category you go to the section part. Here you show several sections (again, go as wide as you need) with again some detail examples. After that: the details layer shows each item. By giving some samples of the underlaying layer you achieve several things: you make the layer attractive by showing several different things, you show some highlights so the user sees actual content and you provide a shortcut to the layers underneath. The image below is borrowed from the http://design.windows.com website which has tons and tons of examples: For our app we’ll use this layout. So what will we show? Well, let’s see what sorts of features our app has to offer. I’ll repeat them here: Note planes Add pictures of that plane Notify friends of new spots Share new spots on social media Write down arrival times Write down departure times Write down the runway they take I am sure you can think of some more items but for now we'll use these. In the hub we’ll show something that represents “Spots”, “Friends”, “Social”. Apparently we have an inner list of spotter-friends that are in the app, while we also have to whole world in social. In the layer below we show something else, depending on what the user choose. When they choose “Spots” we’ll display the last spots, last spots by our friends (so we can actually jump from this category to the one next to it) and so on. When they choose a “spot” (or press the + icon in the App bar, which I’ll talk about next time) they go to the lowest and final level that shows details about that spot, including a picture, date and time and the notes belonging to that entry. You’d be amazed at how easy it is to organize your app this way. If you don’t have enough room in these three layers you probably could easily get away with grouping items. Take a look at our hub: we have three completely different things in one place. If you still can’t fit it all in in a logical and consistent way, chances are you are trying to do too much in this app. Go back to your mission statement, determine if it is specific enough and if your feature list helps that statement or makes it unclear. Go ahead. Give it a go! Next time we’ll talk about the look and feel, the charms and the app-bar….

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  • MVVM Light V4 preview (BL0014) release notes

    - by Laurent Bugnion
    I just pushed to Codeplex an update to the MVVM Light source code. This is an early preview containing some of the features that I want to release later under the version 4. If you find these features useful for your project, please download the source code and build the assemblies. I will appreciate greatly any issue report. This version is labeled “V4.0.0.0/BL0014”. The “BL” string is an old habit that we used in my days at Siemens Building Technologies, called a “base level”. Somehow I like this way of incrementing the “base level” independently of any other consideration (such as alpha, beta, CTP, RTM etc) and continue to use it to tag my software versions. In Microsoft parlance, you could say that this is an early CTP of MVVM Light V4. Caveat The code is unit tested, but as we all know this does not mean that there are no bugs This code has not yet been used in production. Again, your help in testing this is greatly appreciated, so please report all bugs to me! What’s new? The following features have been implemented: Misc Various “maintenance work”. All WPF assemblies (that is .NET35 and .NET4) now allow partially trusted callers. It means that you can use them in am XBAP in partial trust mode. Testing Various test updates Added Windows Phone 7 unit tests Note: For Windows Phone 7, due to an issue in the unit test framework, not all tests can be executed. I had to isolate those tests for the moment. The error was reported to Microsoft. ViewModelBase The constructor is now public to allow serialization (especially useful on the phone to tombstone the state). ViewModelBase.MessengerInstance now returns Messenger.Default unless it is set explicitly. Previously, MessengerInstance was returning null, which was complicating the code. Two new ways to raise the PropertyChanged event have been added. See below for details. Messenger Updated the IMessenger interface with all public members from the Messenger class. Previously some members were missing. A new Unregister method is now available, allowing to unregister a recipient for a given token. RelayCommand RaiseCanExecuteChanged now acts the same in Windows Presentation Foundation than in Silverlight. In previous versions, I was relying on the CommandManager to raise the CanExecuteChanged event in WPF. However, it was found to be too unreliable, and a more direct way of raising the event was found preferable. See below for details. Raising the PropertyChanged event A very much requested update is now included: the ability to raise the PropertyChanged event in a viewmodel without using “magic strings”. Personally, I don’t see strings as a major issue, thanks to two features of the MVVM Light Toolkit: In the DEBUG configuration, every time that the RaisePropertyChanged method is called, the name of the property is checked against all existing properties of the viewmodel. Should the property name be misspelled (because of a typo or refactoring), an exception is thrown, notifying the developer that something is wrong. To avoid impacting the performance, this check is only made in DEBUG configuration, but that should be enough to warn the developers in case they miss a rename. The property name is defined as a public constant in the “mvvminpc” code snippet. This allows checking the property name from another class (for example if the PropertyChanged event is handled in the view). It also allows changing the property name in one place only. However, these two safeguards didn’t satisfy some of the users, who requested another way to raise the PropertyChanged event. In V4, you can now do the following: Using lambdas private int _myProperty; public int MyProperty { get { return _myProperty; } set { if (_myProperty == value) { return; } _myProperty = value; RaisePropertyChanged(() => MyProperty); } } This raises the property changed event using a lambda expression instead of the property name. Light reflection is used to get the name. This supports Intellisense and can easily be refactored. You can also broadcast a PropertyChangedMessage using the Messenger.Default instance with: private int _myProperty; public int MyProperty { get { return _myProperty; } set { if (_myProperty == value) { return; } var oldValue = _myProperty; _myProperty = value; RaisePropertyChanged(() => MyProperty, oldValue, value, true); } } Using no arguments When the RaisePropertyChanged method is called within a setter, you can also omit the property name altogether. This will fail if executed outside of the setter however. Also, to avoid confusion, there is no way to broadcast the PropertyChangedMessage using this syntax. private int _myProperty; public int MyProperty { get { return _myProperty; } set { if (_myProperty == value) { return; } _myProperty = value; RaisePropertyChanged(); } } The old way Of course the “old” way is still supported, without broadcast: public const string MyPropertyName = "MyProperty"; private int _myProperty; public int MyProperty { get { return _myProperty; } set { if (_myProperty == value) { return; } _myProperty = value; RaisePropertyChanged(MyPropertyName); } } And with broadcast: public const string MyPropertyName = "MyProperty"; private int _myProperty; public int MyProperty { get { return _myProperty; } set { if (_myProperty == value) { return; } var oldValue = _myProperty; _myProperty = value; RaisePropertyChanged(MyPropertyName, oldValue, value, true); } } Performance considerations It is notorious that using reflection takes more time than using a string constant to get the property name. However, after measuring for all platforms, I found the differences to be very small. I will measure more and submit the results to the community for evaluation, because some of the results are actually surprising (for example, using the Messenger to broadcast a PropertyChangedMessage does not significantly increase the time taken to raise the PropertyChanged event and update the bindings). For now, I submit this code to you, and would be delighted to hear about your own results. Raising the CanExecuteChanged event manually In WPF, until now, the CanExecuteChanged event for a RelayCommand was raised automatically. Or rather, it was attempted to be raised, using a feature that is only available in WPF called the CommandManager. This class monitors the UI and when something occurs, it queries the state of the CanExecute delegate for all the commands. However, this proved unreliable for the purpose of MVVM: Since very often the value of the CanExecute delegate changes according to non-UI events (for example something changing in the viewmodel or in the model), raising the CanExecuteChanged event manually is necessary. In Silverlight, the CommandManager does not exist, so we had to raise the event manually from the start. This proved more reliable, and I now changed the WPF implementation of the RaiseCanExecuteChanged method to be the exact same in WPF than in Silverlight. For instance, if a command must be enabled when a string property is set to a value other than null or empty string, you can do: public MainViewModel() { MyTestCommand = new RelayCommand( () => DoSomething(), () => !string.IsNullOrEmpty(MyProperty)); } public const string MyPropertyName = "MyProperty"; private string _myProperty = string.Empty; public string MyProperty { get { return _myProperty; } set { if (_myProperty == value) { return; } _myProperty = value; RaisePropertyChanged(MyPropertyName); MyTestCommand.RaiseCanExecuteChanged(); } } Logo update I made a minor change to the logo: Some people found the lack of the word “light” (as in MVVM Light Toolkit) confusing. I thought it was cool, because the feather suggests the idea of lightness, however I can see the point. So I added the word “light” to the logo. Things should be quite clear now. What’s next? This is only the first of a series of releases that will bring MVVM Light to V4. In the next weeks, I will continue to add some very requested features and correct some issues in the code. I will probably continue this fashion of releasing the changes to the public as source code through Codeplex. I would be very interested to hear what you think of that, and to get feedback about the changes. Cheers, Laurent   Laurent Bugnion (GalaSoft) Subscribe | Twitter | Facebook | Flickr | LinkedIn

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  • Help With Database Layout

    - by three3
    Hello everyone, I am working on a site similar to Craigslist where users can make postings and sell items in different cities. One difference between my site and Craigslist will be you will be able to search by zip code instead of having all of the cities listed on the page. I already have the ZIP Code database that has all of the city, state, latitude, longitude, and zip code info for each city. Okay, so to dive into what I need done and what I need help with: 1.) Although I have the ZIP Code database, it is not setup perfectly for my use. (I downloaded it off of the internet for free from http://zips.sourceforge.net/) 2.) I need help setting up my database structure (Ex: How many different tables should I use and how should I link them) I will be using PHP and MySQL. These our my thoughts so far on how the database can be setup: (I am not sure if this will work though.) Scenario: Someone goes to the homepage and it will tell them, "Please enter your ZIP Code.". If they enter "17241" for example, this ZIP Code is for a city named Newville located in Pennsylvania. The query would look like this with the current database setup: SELECT city FROM zip_codes WHERE zip = 17241; The result of the query would be "Newville". The problem I see here now is when they want to post something in the Newville section of the site, I will have to have an entire table setup just for the Newville city postings. There are over 42,000 cities which means I would have to have over 42,000 tables (one for each city) so that would be insane to have to do it that way. One way I was thinking of doing it was to add a column to the ZIP Code database called "city_id" which would be a unique number assigned to each city. So for example, the city Newville would have a city_id of 83. So now if someone comes and post a listing in the city Newville I would only need one other table. That one other table would be setup like this: CREATE TABLE postings ( posting_id INT NOT NULL AUTO_INCREMENT, for_sale LONGTEXT NULL, for_sale_date DATETIME NULL, for_sale_city_id INT NULL, jobs LONGTEXT NULL, jobs_date DATETIME NULL, jobs_city_id INT NULL, PRIMARY KEY(posting_id) ); (The for_sale and job_ column names are categories of the types of postings users will be able to list under. There will be many more categories than just those two but this is just for example.) So now when when someone comes to the website and they are looking for something to buy and not sell, they can enter their ZIP Code, 17241, for example, and this is the query that will run: SELECT city, city_id FROM zip_codes WHERE zip = 17241; //Result: Newville 83 (Please note that I will be using PHP to store the ZIP Code the user enters in SESSIONS and Cookies to remember them throughout the site) Now it will tell them, "Please choose your category.". If they choose the category "Items For Sale" then this is the query to run and sort the results: SELECT posting_id, for_sale, for_sale_date FROM postings WHERE for_sale_city_id = $_SESSION['zip_code']; Will this work? So now my question to everyone is will this work? I am pretty sure it will but I do not want to set this thing up and realize I overlooked something and have to start from all over from scratch. Any opinions and ideas are welcomed and I will listen to anyone who has some thoughts. I really appreciate the help in advance :D

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  • WCF timeouts are a nightmare

    - by Greg
    We have a bunch of WCF services that work almost all of the time, using various bindings, ports, max sizes, etc. The super-frustrating thing about WCF is that when it (rarely) fails, we are powerless to find out why it failed. Sometimes you will get a message that looks like this: System.ServiceModel.CommunicationException: The socket connection was aborted. This could be caused by an error processing your message or a receive timeout being exceeded by the remote host, or an underlying network resource issue. Local socket timeout was '01:00:00'. --- System.IO.IOException: Unable to read data from the transport connection: An existing connection was forcibly closed by the remote host. The problem is that the local socket timeout it's giving you is merely an attempt to be convenient. It may or may not be the cause of the problem. But OK, sometimes networks have issues. No big deal. We can retry or something. But here's the huge problem. On top of failing to tell you which precisely which timeout (if any) resulted in the failure ("your server-side receive timeout was exceeded," or something, would be helpful), WCF seems to have two types of timeouts. Timeout Type #1) A timeout, that, if increased, would increase the chance of your operation's success. So, the pertinent timeout is an hour, you are uploading a huge file that will take an hour and twenty minutes. It fails. You increase the timeout, it succeeds. I have no no problem with this type of timeout. Timeout Type #2) A timeout which merely defines how long you have to wait for the service to actually fail and give you an error, but modifying the value of this timeout has no impact on the chance of success. Basically, something happens during the first second of the service request which mucks things up. It will never recover. WCF doesn't magically retry the network connection for you. Fine, sometimes establishing a network connection doesn't go well. But, if your timeout is 2 hours, you have to wait 2 whole hours with no chance of it ever working before it finally acknowledges that it didn't work and gives you the error. But the error you see in both cases looks the same. With timeout Type #2, it still looks like you are running into a timeout. But, you could increase all of your timeouts to 4 years, and all it would do is make it take 4 years to get an error message. I know that Type #2 exists because I can do an operation that is known to complete in less than a minute when successful, and have it take 2 hours to fail. But, if I kill it and retry, it succeeds quickly. (If you are wondering why there might be a 2 hour timeout on an operation that takes less than a minute, there are times I run the operation with a much larger file and it could take over an hour.) So, to combat the problem with Type #2, you'd want your timeout to be really quick so you immediately know if there is a problem. Then you can retry. But the insurmountable problem is that because I don't know which timeouts are the cause of failure, I don't know what timeouts are Type #1 and which ones are Type #2. There may be one timeout (let's say the client-side send timeout) that acts like Type #1 in some cases and Type #2 in others. I have no idea, and I have no way of finding out. Does anyone know how to track down Type #2 timeouts so I can set them to low values without having to shorten actual (read: Type #1) timeouts and lower the chance of success? Thank you.

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  • Small performance test on a web service

    - by vtortola
    Hi, I'm trying to develop a small application that test how many requests per second can my service support but I think I'm doing something wrong. The service is in an early development stage, but I'd like to have this test handy in order to check in time to time I'm not doing something that decrease the performance. The problem is that I cannot get the web server or the database server go to the 100% of CPU. I'm using three different computers, in one is the web server (WinSrv Standard 2008 x64 IIS7), in other the database (Win 2K - SQL Server 2005) and the last is my computer (Win7 x64 ultimate), where I'll run the test. The computers are connected through a 100 ethernet switch. The request POST is 9 bytes and the response will be 842 bytes. The test launches several threads, and each thread has a "while" loop, in each loop it creates a WebRequest object, performs a call, increment a common counter and waits between 1 and 5 millisencods, then it do it again: static Int32 counter = 0; static void Main(string[] args) { ServicePointManager.DefaultConnectionLimit = 250; Console.WriteLine("Ready. Press any key..."); Console.ReadKey(); Console.WriteLine("Running..."); String localhost = "localhost"; String linuxmono = "192.168.1.74"; String server= "192.168.1.5:8080"; DateTime start = DateTime.Now; Random r = new Random(DateTime.Now.Millisecond); for (int i = 0; i < 50; i++) { new Thread(new ParameterizedThreadStart(Test)).Start(server); Thread.Sleep(r.Next(1, 3)); } Thread.Sleep(2000); while (true) { Console.WriteLine("Request per second :" + counter/DateTime.Now.Subtract(start).TotalSeconds ); Thread.Sleep(3000); } } public static void Test(Object ip) { Guid guid = Guid.NewGuid(); Random r = new Random(DateTime.Now.Millisecond); while (true) { String test = "<lalala/>"; WebRequest req = WebRequest.Create("http://" + (String)ip + "/WebApp/"+guid.ToString()+"/Data/Tables=whatever"); req.Method = "POST"; req.ContentType = "application/xml"; req.Credentials = new NetworkCredential("aaa", "aaa","domain"); Byte[] array = Encoding.UTF8.GetBytes(test); req.ContentLength = array.Length; using (Stream reqStream = req.GetRequestStream()) { reqStream.Write(array, 0, array.Length); reqStream.Close(); } using (Stream responseStream = req.GetResponse().GetResponseStream()) { String response = new StreamReader(responseStream).ReadToEnd(); if (response.Length != 842) Console.Write(" EEEE "); } Interlocked.Increment(ref counter); Thread.Sleep(r.Next(1,5)); } } If I run the test neither of the computers do an excesive CPU usage. Let's say I get a X requests per second, if I run the console application two times at the same moment, I get X/2 request per second in each one... but still the web server is on 30% of CPU, the database server on 25%... I've tried to remove the thread.sleep in the loop, but it doesn't make a big difference. I'd like to put the machines to the maximun, to check how may requests per second they can provide. I guessed that I could do it in this way... but apparently I'm missing something here... What is the problem? Kind regards.

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  • Resolving html entities with NSXMLParse on iPhone

    - by Roberto
    Hi all, i think i read every single web page relating to this problem but i still cannot find a solution to it, so here i am. I have an HTML web page wich is not under my control and i need to parse it from my iPhone application. Here it is a sample of the web page i'm talking about: <HTML> <HEAD> <META http-equiv="Content-Type" content="text/html; charset=ISO-8859-1"> </HEAD> <BODY> <LI class="bye bye" rel="hello 1"> <H5 class="onlytext"> <A name="morning_part">morning</A> </H5> <DIV class="mydiv"> <SPAN class="myclass">something about you</SPAN> <SPAN class="anotherclass"> <A href="http://www.google.it">Bye Bye &egrave; un saluto</A> </SPAN> </DIV> </LI> </BODY> </HTML> I'm using NSXMLParser and it is going well till it find the è html entity. It calls foundCharacters: for "Bye Bye" and then it calls resolveExternalEntityName:systemID:: with an entityName of "egrave". In this method i'm just returning the character "è" trasformed in an NSData, the foundCharacters is called again adding the string "è" to the previous one "Bye Bye " and then the parser raise the NSXMLParserUndeclaredEntityError error. I have no DTD and i cannot change the html file i'm parsing. Do you have any ideas on this problem? Thanks in advance to all of you, Rob. Update (12/03/2010). After the suggestion of Griffo i ended up with something like this: data = [self replaceHtmlEntities:data]; NSXMLParser *parser = [[NSXMLParser alloc] initWithData:data]; [parser setDelegate:self]; [parser parse]; where replaceHtmlEntities:(NSData *) is something like this: - (NSData *)replaceHtmlEntities:(NSData *)data { NSString *htmlCode = [[NSString alloc] initWithData:data encoding:NSISOLatin1StringEncoding]; NSMutableString *temp = [NSMutableString stringWithString:htmlCode]; [temp replaceOccurrencesOfString:@"&amp;" withString:@"&" options:NSLiteralSearch range:NSMakeRange(0, [temp length])]; [temp replaceOccurrencesOfString:@"&nbsp;" withString:@" " options:NSLiteralSearch range:NSMakeRange(0, [temp length])]; ... [temp replaceOccurrencesOfString:@"&Agrave;" withString:@"À" options:NSLiteralSearch range:NSMakeRange(0, [temp length])]; NSData *finalData = [temp dataUsingEncoding:NSISOLatin1StringEncoding]; return finalData; } But i am still looking the best way to solve this problem. I will try TouchXml in the next days but i still think that there should be a way to do this using NSXMLParser API, so if you know how, feel free to write it here :)

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  • Fixing predicated NSFetchedResultsController/NSFetchRequest performance with SQLite backend?

    - by Jaanus
    I have a series of NSFetchedResultsControllers powering some table views, and their performance on device was abysmal, on the order of seconds. Since it all runs on main thread, it's blocking my app at startup, which is not great. I investigated and turns out the predicate is the problem: NSPredicate *somePredicate = [NSPredicate predicateWithFormat:@"ANY somethings == %@", something]; [fetchRequest setPredicate:somePredicate]; I.e the fetch entity, call it "things", has a many-to-many relation with entity "something". This predicate is a filter that limits the results to only things that have a relation with a particular "something". When I removed the predicate for testing, fetch time (the initial performFetch: call) dropped (for some extreme cases) from 4 seconds to around 100ms or less, which is acceptable. I am troubled by this, though, as it negates a lot of the benefit I was hoping to gain with Core Data and NSFRC, which otherwise seems like a powerful tool. So, my question is, how can I optimize this performance? Am I using the predicate wrong? Should I modify the model/schema somehow? And what other ways there are to fix this? Is this kind of degraded performance to be expected? (There are on the order of hundreds of <1KB objects.) EDIT WITH DETAILS: Here's the code: [fetchRequest setFetchLimit:200]; NSLog(@"before fetch"); BOOL success = [frc performFetch:&error]; if (!success) { NSLog(@"Fetch request error: %@", error); } NSLog(@"after fetch"); Updated logs (previously, I had some application inefficiencies degrading the performance here. These are the updated logs that should be as close to optimal as you can get under my current environment): 2010-02-05 12:45:22.138 Special Ppl[429:207] before fetch 2010-02-05 12:45:22.144 Special Ppl[429:207] CoreData: sql: SELECT DISTINCT 0, t0.Z_PK, t0.Z_OPT, <model fields> FROM ZTHING t0 LEFT OUTER JOIN Z_1THINGS t1 ON t0.Z_PK = t1.Z_2THINGS WHERE t1.Z_1SOMETHINGS = ? ORDER BY t0.ZID DESC LIMIT 200 2010-02-05 12:45:22.663 Special Ppl[429:207] CoreData: annotation: sql connection fetch time: 0.5094s 2010-02-05 12:45:22.668 Special Ppl[429:207] CoreData: annotation: total fetch execution time: 0.5240s for 198 rows. 2010-02-05 12:45:22.706 Special Ppl[429:207] after fetch If I do the same fetch without predicate (by commenting out the two lines in the beginning of the question): 2010-02-05 12:44:10.398 Special Ppl[414:207] before fetch 2010-02-05 12:44:10.405 Special Ppl[414:207] CoreData: sql: SELECT 0, t0.Z_PK, t0.Z_OPT, <model fields> FROM ZTHING t0 ORDER BY t0.ZID DESC LIMIT 200 2010-02-05 12:44:10.426 Special Ppl[414:207] CoreData: annotation: sql connection fetch time: 0.0125s 2010-02-05 12:44:10.431 Special Ppl[414:207] CoreData: annotation: total fetch execution time: 0.0262s for 200 rows. 2010-02-05 12:44:10.457 Special Ppl[414:207] after fetch 20-fold difference in times. 500ms is not that great, and there does not seem to be a way to do it in background thread or otherwise optimize that I can think of. (Apart from going to a binary store where this becomes a non-issue, so I might do that. Binary store performance is consistently ~100ms for the above 200-object predicated query.) (I nested another question here previously, which I now moved away).

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  • How to get objects to react to touches in Cocos2D?

    - by Wayfarer
    Alright, so I'm starting to learn more about Coco2D, but I'm kinda frusterated. A lot of the tutorials I have found are for outdated versions of the code, so when I look through and see how they do certain things, I can't translate it into my own program, because a lot has changed. With that being said, I am working in the latest version of Coco2d, version 0.99. What I want to do is create a sprite on the screen (Done) and then when I touch that sprite, I can have "something" happen. For now, let's just make an alert go off. Now, I got this code working with the help of a friend. Here is the header file: // When you import this file, you import all the cocos2d classes #import "cocos2d.h" // HelloWorld Layer @interface HelloWorld : CCLayer { CGRect spRect; } // returns a Scene that contains the HelloWorld as the only child +(id) scene; @end And here is the implementation file: // // cocos2d Hello World example // http://www.cocos2d-iphone.org // // Import the interfaces #import "HelloWorldScene.h" #import "CustomCCNode.h" // HelloWorld implementation @implementation HelloWorld +(id) scene { // 'scene' is an autorelease object. CCScene *scene = [CCScene node]; // 'layer' is an autorelease object. HelloWorld *layer = [HelloWorld node]; // add layer as a child to scene [scene addChild: layer]; // return the scene return scene; } // on "init" you need to initialize your instance -(id) init { // always call "super" init // Apple recommends to re-assign "self" with the "super" return value if( (self=[super init] )) { // create and initialize a Label CCLabel* label = [CCLabel labelWithString:@"Hello World" fontName:@"Times New Roman" fontSize:64]; // ask director the the window size CGSize size = [[CCDirector sharedDirector] winSize]; // position the label on the center of the screen label.position = ccp( size.width /2 , size.height/2 ); // add the label as a child to this Layer [self addChild: label]; CCSprite *sp = [CCSprite spriteWithFile:@"test2.png"]; sp.position = ccp(300,200); [self addChild:sp]; float w = [sp contentSize].width; float h = [sp contentSize].height; CGPoint aPoint = CGPointMake([sp position].x - (w/2), [sp position].y - (h/2)); spRect = CGRectMake(aPoint.x, aPoint.y, w, h); CCSprite *sprite2 = [CCSprite spriteWithFile:@"test3.png"]; sprite2.position = ccp(100,100); [self addChild:sprite2]; //[self registerWithTouchDispatcher]; self.isTouchEnabled = YES; } return self; } // on "dealloc" you need to release all your retained objects - (void) dealloc { // in case you have something to dealloc, do it in this method // in this particular example nothing needs to be released. // cocos2d will automatically release all the children (Label) // don't forget to call "super dealloc" [super dealloc]; } - (void)ccTouchesEnded:(NSSet *)touches withEvent:(UIEvent *)event { UITouch *touch = [touches anyObject]; //CGPoint location = [[CCDirector sharedDirector] convertCoordinate:[touch locationInView:touch.view]]; CGPoint location = [touch locationInView:[touch view]]; location = [[CCDirector sharedDirector] convertToGL:location]; if (CGRectContainsPoint(spRect, location)) { UIAlertView *alert = [[UIAlertView alloc] initWithTitle:@"Win" message:@"testing" delegate:nil cancelButtonTitle:@"okay" otherButtonTitles:nil]; [alert show]; [alert release]; NSLog(@"TOUCHES"); } NSLog(@"Touch got"); } However, this only works for 1 object, the sprite which I create the CGRect for. I can't do it for 2 sprites, which I was testing. So my question is this: How can I have all sprites on the screen react to the same event when touched? For my program, the same event needs to be run for all objects of the same type, so that should make it a tad easier. I tried making a subclass of CCNode and over write the method, but that just didn't work at all... so I'm doing something wrong. Help would be appreciated!

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  • Token replacement

    - by ClarkeyBoy
    Hey, I currently have a system on my website whereby I can put something like "[cfe]" anywhere in the site and, when the page is rendered, it will replace it with the root to the customer front end (same for "[afe]" and admin front end - so in the admin front end I can put "[cfe]/Default.aspx" to link to the homepage on the customer front end. This is in place as I have a development version of the site, then a test and a live version too. All 3 may have different roots to each section (for example the way the website is set up, the root to the admin front end in test is "/test/Administration/", but in live and development it is just "/Administration/"). Which version it is depends on the URL - all my development sites are in a folder called "development", whereas test is in a folder called "test" and any live urls do not contain either of these. There are also 3 different databases - one for each. All 3, obviously, require a different connection string. I also have a string replacement function in place which can change, for example, "[Product:Cards]" to point to the Cards catalogue page. Problem is that for this I go through all the products and do a replacement on "[Product:" & Product.Name() & "]". However I would like to take this further. I would like to pick up these custom strings when the page is rendered so it picks up "[Product:Cards]" and then goes off to find product "Cards" and replaces the string with a link to the Cards page, rather than looping through all the products and doing a replace just on the off chance that there are any replacements to make. One use for this, which I may start using in the future if I can figure out how to do this, is like on Wikipedia where you put the title of the page you want to point to, then a divider (think its a pipe from memory) then the link text. I would like to apply this to the above situation. This way broken links can also be picked up, and reported to admin (a major advantage as they can then locate them and remove the link or add the product / page that it refers to). I would like to take this to the stage where content of entire pages can be rearranged (kinda like web parts, but not as advanced as that). I mean like so you can put [layout type="3columnImageTopRight" image="imageurl"]Content here[/layout]. This will display, as specified, an image in the top right (with padding at the left and bottom) and 3 columns - maybe with the image spanning one or two columns). The imageurl can be specified as another token: maybe like [Image:imagename.gif] or something. This replaces it with the root to the image folder and then the specified filename. I have not really looked into how I am going to split the content into 3 columns yet, but this would be something to look at for my dissertation and then implement after my project deadline at least. Does anyone have any ideas or pointers which could help me with this? Also if this is not strictly token replacement then please point me to what it is, so I can further develop this. Thanks in advance, Regards, Richard Clarke

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  • How to Access a descendant object's internal method in C#

    - by Giovanni Galbo
    I'm trying to access a method that is marked as internal in the parent class (in its own assembly) in an object that inherits from the same parent. Let me explain what I'm trying to do... I want to create Service classes that return IEnumberable with an underlying List to non-Service classes (e.g. the UI) and optionally return an IEnumerable with an underlying IQueryable to other services. I wrote some sample code to demonstrate what I'm trying to accomplish, shown below. The example is not real life, so please remember that when commenting. All services would inherit from something like this (only relevant code shown): public class ServiceBase<T> { protected readonly ObjectContext _context; protected string _setName = String.Empty; public ServiceBase(ObjectContext context) { _context = context; } public IEnumerable<T> GetAll() { return GetAll(false); } //These are not the correct access modifiers.. I want something //that is accessible to children classes AND between descendant classes internal protected IEnumerable<T> GetAll(bool returnQueryable) { var query = _context.CreateQuery<T>(GetSetName()); if(returnQueryable) { return query; } else { return query.ToList(); } } private string GetSetName() { //Some code... return _setName; } } Inherited services would look like this: public class EmployeeService : ServiceBase<Employees> { public EmployeeService(ObjectContext context) : base(context) { } } public class DepartmentService : ServiceBase<Departments> { private readonly EmployeeService _employeeService; public DepartmentService(ObjectContext context, EmployeeService employeeService) : base(context) { _employeeService = employeeService; } public IList<Departments> DoSomethingWithEmployees(string lastName) { //won't work because method with this signature is not visible to this class var emps = _employeeService.GetAll(true); //more code... } } Because the parent class lives is reusable, it would live in a different assembly than the child services. With GetAll(bool returnQueryable) being marked internal, the children would not be able to see each other's GetAll(bool) method, just the public GetAll() method. I know that I can add a new internal GetAll method to each service (or perhaps an intermediary parent class within the same assembly) so that each child service within the assembly can see each other's method; but it seems unnecessary since the functionality is already available in the parent class. For example: internal IEnumerable<Employees> GetAll(bool returnIQueryable) { return base.GetAll(returnIQueryable); } Essentially what I want is for services to be able to access other service methods as IQueryable so that they can further refine the uncommitted results, while everyone else gets plain old lists. Any ideas? EDIT You know what, I had some fun playing a little code golf with this... but ultimately I wouldn't be able to use this scheme anyway because I pass interfaces around, not classes. So in my example GetAll(bool returnIQueryable) would not be in the interface, meaning I'd have to do casting, which goes against what I'm trying to accomplish. I'm not sure if I had a brain fart or if I was just too excited trying to get something that I thought was neat to work. Either way, thanks for the responses.

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