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  • Poll on Entity Framework 4 &ndash; one year on

    - by Eric Nelson
    12 months back (today is March 15th 2010) on the 16th of  March 2009 I created a poll on Entity Framework v1 – the marmite of ORMs? A quick poll…. Entity Framework v1 was getting a mixed reception at the time – I met developers who genuinely hated it and I met developers who were loving the productivity improvements they were seeing. There were definitely issues with v1, too many IMHO. Which is why the product team placed a huge effort on listening to the community to drive the feature set for v2 (which ultimately was named Entity Framework 4 as it ships with .NET 4). I think overall the team have done a great job. It isn’t perfect in .NET 4 (which is why the team are busy on post .NET 4 improvements) but I would happily use it and recommend it for a wide variety of projects – much wider than I would have with v1. I am speaking on EF 4 at www.devweek.com this Wednesday and I thought it would be fun to put a new version of the poll out and see how v4 is being received. Obviously the big difference is we have not yet shipped EF4 vs when I did the original poll on EF1. March 2010 poll – please vote Summary of March 2009 poll – it was a tie between positive and negative Total votes 150 Positive about EF v1 42 (15 + 19 + 8) Negative about EF v1  43 (34 + 9)

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  • Advice on how to build html5 basic tile game (multi player, cross device)

    - by Eric
    I just read http://buildnewgames.com/real-time-multiplayer/ which explains the fundamentals and bets practices to build a massive real time multiplayer html5 game. My question is however given the “simplicity” of the game I need to build (simple kind of scratch game where you find or not something behind a tile), do I really need complex tools (canvas or node.js for example) ? The game The gamestakes place with a picture of our office as a background (tilemap). For HR purpose, we wish to create the following game fore employees: each day they can come to the website and click on a certain number of tiles (3 max per day) and find behind it motivation advice and interesting facts about the company. The constraints and rules the screen is divided into isometric 2D square tiles. There are basically an image (photograph of our office) number of tiles on the screen game: about 10,000 to much more (with scroll , see below) the players can scroll in 4 directions there are only 2 types of tiles: already open and closed player can open tiles that have not been yet open by other players there is no path for players : any player can click on any tile on the screen at any moment (if it’s not already done by another player) 2 players can’t be on the same tile at the same moment (or if they can, I’ll have to manage to see which one clicked on it first) only one type of player (all with similar roles), no weapon, no internal score… very simple game. no complex physics (collision only occurs if 2 players are on the same tile) The target I need to achieve: cross device, cross browsers high performance reaction (subsecond reactions) average nb of players per hour: up to 10K players per hour (quite high indeed but it’s because we aim at proving our case for the game to our business unit) So what I would like to know: 2D Tiled map: Do I need tiledmapeditor or can I enable me split the screen like here ? should I use canvas or plain html/css could be sufficient for my need? do I need a game engine/framework such as melon.js or crafty./js ? (even if the game play is extremely basic, I do need mouse and touché device support, mouse emulations on touch devices…) or ca I easily/quickly do it without? for my constraints and targets, should I use CPU acceleration ? for my constraints and targets, should I use web workers ? for the database, for a massively real time game should I avoid to put the current locations of player in MySQL as i feel it might slow me down. What kind of DB should I implement ? Thanks for your help !

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  • Why can't I mount the Ubuntu 12.04 installer ISOs in Mac OS X?

    - by eric
    Over the past few days, I have downloaded both the 32 and 64 bit version of server and desktop to install on an Intel based PC. It is normal from within OS X to double click on an ISO and it will mount the ISO in the finder as well as within disk utility. When I attempt to mount any of the Ubuntu ISOs I downloaded, OS X returns the error message The following disk images couldn't be opened. The reason given for the error is no mountable filesystem. However, I am still able to open the ISO directly from disk utility and burn it to a DVD/CD. What has changed in this release to cause this? Is there something wrong with the current ISOs? The OS X machine I am using is only two weeks old and is having no issues with any other ISO.

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  • How should I implement the repository pattern for complex object models?

    - by Eric Falsken
    Our data model has almost 200 classes that can be separated out into about a dozen functional areas. It would have been nice to use domains, but the separation isn't that clean and we can't change it. We're redesigning our DAL to use Entity Framework and most of the recommendations that I've seen suggest using a Repository pattern. However, none of the samples really deal with complex object models. Some implementations that I've found suggest the use of a repository-per-entity. This seems ridiculous and un-maintainable for large, complex models. Is it really necessary to create a UnitOfWork for each operation, and a Repository for each entity? I could end up with thousands of classes. I know this is unreasonable, but I've found very little guidance implementing Repository, Unit Of Work, and Entity Framework over complex models and realistic business applications.

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  • Cannot get grub menu to timeout (or go away)

    - by Eric
    I am running Ubuntu 12.04. I cannot for the life of me get the grub menu (with options) to go away. I would like it to auto-boot into the first option. I've edited /etc/default/grub so that it looks like the following: GRUB_DEFAULT=0 GRUB_HIDDEN_TIMEOUT=0 GRUB_HIDDEN_TIMEOUT_QUIET=true GRUB_TIMEOUT=10 GRUB_DISTRIBUTOR=`lsb_release -i -s 2> /dev/null || echo Debian` GRUB_CMDLINE_LINUX_DEFAULT="quiet splash" GRUB_CMDLINE_LINUX="" After this, I ran sudo update-grub. Then I realized that I had grub 2, so I ran sudo update-grub2 (both make it look like they create the grub.cfg file successfully. I restart my computer and it goes back to the grub menu and just sits there until I make a choice. This is not desirable if the power goes out... :-) I also tried reinstalling grub using sudo grub-install /dev/sda - also unsuccessful

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  • Syncing between computers / batch download

    - by Eric
    I have synced files from two different devices. Now I want to copy to contents of the synced files to the other device. Using the web interface, it seems only possible to download one file at a time. Is there a way to setup automatic sync or to download all the files in a folder with one step? Thanks in advance for your time. I was using the web interface to ubuntu, but an answer for Ubuntu One would also be ok.

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  • Best way to 'harden' embedded ext4 file server against unexpected loss of power?

    - by Jeremy Friesner
    Hi all, First, a little background: my company makes an audio streaming device that is a headless, rack-mounted Linux box with a couple of SSDs attached. Each SSD is formatted with ext4. The users can connect to the system using Samba/CIFS to upload new audio files or access existing ones. There is also custom software for streaming out audio over the network. This is all fine. The only problem is that the users are audio people, not computer people, and see the system as a 'black box', not as a computer. Which means that at the end of the day, they aren't going to ssh in to the box and enter "/sbin/shutdown -h"; they are just going to cut power to the rack and leave, and expect things to still work properly the next day. Since ext4 has journalling, journal checksumming, etc, this mostly works. The only time it doesn't work is when someone uploads a new file via Samba and then cuts power to the system before the uploaded data has been fully flushed to the disk. In that case, they come in the next day and find that their new file has been truncated or is missing entirely, and are unhappy. My question is, what is the best way to avoid this problem? Is there a way to get smbd to call "sync" at the end of every upload? (Performance on uploads isn't so important, since they only happen occasionally). Or is there a way to tell ext4 to automatically flush within a few seconds of any change to a file? (Again, performance can be sacrificed for safety here) Should I set a particular write-ordering mode, activate barriers, etc?

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  • Getting old bluetooth headset out of standby on Windows 7

    - by luagh45
    I think this is the problem, and if not, I'm open to suggestions. I have an old Jabra BT200 that I used to use on my phone. When a phone call was coming in it would beep using its own noises (meaning the phone never rang inside the headset) and then I could push the 'answer/hang up' button and sound and mic would start working. I have now paired it with Windows 7, and it looks good. Under the playback menu I have 'Bluetooth Hands free Audio / Jabra BT200 (Mono Audio) / Ready', and under the recording menu I have 'Bluetooth Audio Input Device / Jabra BT200 (Mono Audio) / Ready'. However when I try to test the speakers Windows sends a sound, but I never hear it, and when I talk in the mic, Windows never hears me. If I right click either the Bluetooth mic or speakers there's an option to 'Connect', but it's grayed out and I cannot click it. As the final piece of knowledge I have, my headset blinks once every 3 seconds when it's in standby and I can't get that to change. If everything was working it should blink once every second at which point I think all of my problems would be fixed. Hence my issue: I can't seem to get my headset out of standby. On my headset I've tried sending it test noises and then pressing the 'Answer' button, but still nothing. The headset beeps when I press it, so it works, it just doesn't ever come out of standby. Is there maybe some way to trick my headset into thinking it's getting a phone call from my computer?

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  • AVConv increases song duration when converting MP3

    - by chauffch
    I am struggling with the following issue. I want to convert an MP3 ADTS into pure a MP3. I am using AVConv on Ubuntu 12.10. The outcome is a file that has the same size, but the duration is now longer. $ ls -l total 6436 -rw-r--r-- 1 teuf teuf 6586514 nov. 25 09:25 Blindsided_Bon_Iver.mpga $ file Blindsided_Bon_Iver.mpga Blindsided_Bon_Iver.mpga: MPEG ADTS, layer III, v1, 160 kbps, 44.1 kHz, JntStereo $ avconv -i Blindsided_Bon_Iver.mpga -c copy Blindsided_Bon_Iver.mp3 avconv version 0.8.4-4:0.8.4-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the Libav developers built on Nov 6 2012 16:50:25 with gcc 4.6.3 [mp3 @ 0x8c6e240] max_analyze_duration reached Input #0, mp3, from 'Blindsided_Bon_Iver.mpga': Duration: 00:05:29.29, start: 0.000000, bitrate: 160 kb/s Stream #0.0: Audio: mp3, 44100 Hz, stereo, s16, 160 kb/s Output #0, mp3, to 'Blindsided_Bon_Iver.mp3': Metadata: TSSE : Lavf53.21.0 Stream #0.0: Audio: libmp3lame, 44100 Hz, stereo, 160 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Press ctrl-c to stop encoding size= 6432kB time=329.30 bitrate= 160.0kbits/s video:0kB audio:6432kB global headers:0kB muxing overhead 0.002080% $ ls -l total 12868 -rw-rw-r-- 1 teuf teuf 6586129 nov. 27 22:26 Blindsided_Bon_Iver.mp3 -rw-r--r-- 1 teuf teuf 6586514 nov. 25 09:25 Blindsided_Bon_Iver.mpga $ file Blindsided_Bon_Iver.mp3 Blindsided_Bon_Iver.mp3: Audio file with ID3 version 2.4.0, contains: MPEG ADTS, layer III, v1, 32 kbps, 44.1 kHz, Stereo Amarok shows the new file has a duration of 25:27 and has a lot of silence. Am I using an incorrect option? Is it a bug in AVConv? Any ideas how to fix it?

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  • How to embed/hardcode SRT subtitles into mp4 videos with VLC?

    - by Jens Bannmann
    I'm looking for a way to "burn in" or render/rembed/hardcode subtitles (from an SRT file) into an MP4 video with VLC. But no matter what options I use, it never works properly. I get a file that plays video way too fast (audio is normal), or one that plays normally, but actually does not have embedded subtitles. Also, with some options (like the one below) it does not play in QuickTime, only in VLC. So the main question is: how can I make this work in VLC? Secondary questions are: How do I decide which options I should set? Which settings are best if I want to leave the file bitrate etc. the same as much as possible, only embed subtitles? It seems I cannot leave the field empty or Video/Audio unchecked, so I guess I would first need to figure out the original audio and video bitrate. What do the "Scale" and "Channels" options mean? ... none of which are answered within the VLC documentation. For example, this is one set of options I used in the "Advanced Open File…" dialog: Advanced Open File… myFileName.mp4 [ ] Treat as a pipe rather than as a file [x] Load subtitles file: mySubtitleFileName.srt [ ] Play another media synchronously [x] Streaming/Saving Streaming and Transcoding Options [ ] Display the stream locally (o) File [outputFileName.mp4 ] [ ] Dump raw input Encapsulation Method: (MPEG 4 ) Transcoding options [x] Video (mp4v ) Bitrate (kb/s) [256 ] Scale [1 ] [x] Audio (mp3 ) Bitrate (kb/s) [128 ] Channels [1 ]

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  • Getting old bluetooth headset out of standby on Windows 7

    - by luagh45
    I think this is the problem, and if not, I'm open to suggestions. I have an old Jabra BT200 that I used to use on my phone. When a phone call was coming in it would beep using its own noises (meaning the phone never rang inside the headset) and then I could push the 'answer/hang up' button and sound and mic would start working. I have now paired it with Windows 7, and it looks good. Under the playback menu I have 'Bluetooth Hands free Audio / Jabra BT200 (Mono Audio) / Ready', and under the recording menu I have 'Bluetooth Audio Input Device / Jabra BT200 (Mono Audio) / Ready'. However when I try to test the speakers Windows sends a sound, but I never hear it, and when I talk in the mic, Windows never hears me. If I right click either the Bluetooth mic or speakers there's an option to 'Connect', but it's grayed out and I cannot click it. As the final piece of knowledge I have, my headset blinks once every 3 seconds when it's in standby and I can't get that to change. If everything was working it should blink once every second at which point I think all of my problems would be fixed. Hence my issue: I can't seem to get my headset out of standby. On my headset I've tried sending it test noises and then pressing the 'Answer' button, but still nothing. The headset beeps when I press it, so it works, it just doesn't ever come out of standby. Is there maybe some way to trick my headset into thinking it's getting a phone call from my computer?

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  • Running computer from separate room

    - by Dan
    I want my computer to be in the basement, but to use it on the first floor. Which cables should I run through the floor? Can some be wireless or other methods? Here are some of the options I've thought of: Basic: Run DVI, usb (mouse), usb (keyboard), and audio cable (4 cables) USB Hub option: Run DVI, and 1 usb, then using a usb hub split it into mouse, keyboard, maybe even audio (2-3 cables) HDMI Option: If I get a new video card and monitor that supports HDMI, would I be able to run both audio and video through it? Would the monitor have to have an audio out? Also there is a lot of extra bandwidth in the HDMI cables, could I send two monitors on 1 cable or would I have to use 2 cables? How about sending mouse/keyboard through the HDMI cable? I see a lot of monitors with USB hubs built in, but I assume I'd still have to wire HDMI + 1 USB cable to use the USB hubs? X Terminal Machine/Thin Client: I don't really know much about this option. Not sure if it would allow me to run graphics acceleration and watch movies, does anyone know more details about what this would allow me to do? Other options: Any other ways to do this? Can any of this be wireless?

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  • Lync Server 2010

    - by ManojDhobale
    Microsoft Lync Server 2010 communications software and its client software, such as Microsoft Lync 2010, enable your users to connect in new ways and to stay connected, regardless of their physical location. Lync 2010 and Lync Server 2010 bring together the different ways that people communicate in a single client interface, are deployed as a unified platform, and are administered through a single management infrastructure. Workload Description IM and presence Instant messaging (IM) and presence help your users find and communicate with one another efficiently and effectively. IM provides an instant messaging platform with conversation history, and supports public IM connectivity with users of public IM networks such as MSN/Windows Live, Yahoo!, and AOL. Presence establishes and displays a user’s personal availability and willingness to communicate through the use of common states such as Available or Busy. This rich presence information enables other users to immediately make effective communication choices. Conferencing Lync Server includes support for IM conferencing, audio conferencing, web conferencing, video conferencing, and application sharing, for both scheduled and impromptu meetings. All these meeting types are supported with a single client. Lync Server also supports dial-in conferencing so that users of public switched telephone network (PSTN) phones can participate in the audio portion of conferences. Conferences can seamlessly change and grow in real time. For example, a single conference can start as just instant messages between a few users, and escalate to an audio conference with desktop sharing and a larger audience instantly, easily, and without interrupting the conversation flow. Enterprise Voice Enterprise Voice is the Voice over Internet Protocol (VoIP) offering in Lync Server 2010. It delivers a voice option to enhance or replace traditional private branch exchange (PBX) systems. In addition to the complete telephony capabilities of an IP PBX, Enterprise Voice is integrated with rich presence, IM, collaboration, and meetings. Features such as call answer, hold, resume, transfer, forward and divert are supported directly, while personalized speed dialing keys are replaced by Contacts lists, and automatic intercom is replaced with IM. Enterprise Voice supports high availability through call admission control (CAC), branch office survivability, and extended options for data resiliency. Support for remote users You can provide full Lync Server functionality for users who are currently outside your organization’s firewalls by deploying servers called Edge Servers to provide a connection for these remote users. These remote users can connect to conferences by using a personal computer with Lync 2010 installed, the phone, or a web interface. Deploying Edge Servers also enables you to federate with partner or vendor organizations. A federated relationship enables your users to put federated users on their Contacts lists, exchange presence information and instant messages with these users, and invite them to audio calls, video calls, and conferences. Integration with other products Lync Server integrates with several other products to provide additional benefits to your users and administrators. Meeting tools are integrated into Outlook 2010 to enable organizers to schedule a meeting or start an impromptu conference with a single click and make it just as easy for attendees to join. Presence information is integrated into Outlook 2010 and SharePoint 2010. Exchange Unified Messaging (UM) provides several integration features. Users can see if they have new voice mail within Lync 2010. They can click a play button in the Outlook message to hear the audio voice mail, or view a transcription of the voice mail in the notification message. Simple deployment To help you plan and deploy your servers and clients, Lync Server provides the Microsoft Lync Server 2010, Planning Tool and the Topology Builder. Lync Server 2010, Planning Tool is a wizard that interactively asks you a series of questions about your organization, the Lync Server features you want to enable, and your capacity planning needs. Then, it creates a recommended deployment topology based on your answers, and produces several forms of output to aid your planning and installation. Topology Builder is an installation component of Lync Server 2010. You use Topology Builder to create, adjust and publish your planned topology. It also validates your topology before you begin server installations. When you install Lync Server on individual servers, the installation program deploys the server as directed in the topology. Simple management After you deploy Lync Server, it offers the following powerful and streamlined management tools: Active Directory for its user information, which eliminates the need for separate user and policy databases. Microsoft Lync Server 2010 Control Panel, a new web-based graphical user interface for administrators. With this web-based UI, Lync Server administrators can manage their systems from anywhere on the corporate network, without needing specialized management software installed on their computers. Lync Server Management Shell command-line management tool, which is based on the Windows PowerShell command-line interface. It provides a rich command set for administration of all aspects of the product, and enables Lync Server administrators to automate repetitive tasks using a familiar tool. While the IM and presence features are automatically installed in every Lync Server deployment, you can choose whether to deploy conferencing, Enterprise Voice, and remote user access, to tailor your deployment to your organization’s needs.

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  • Sound issues after trying everything

    - by Lerp
    I cannot get my sound working properly, no matter what I do, there's always some problem. It's very annoying as it's the only thing preventing me from making Ubuntu my main OS. At the moment my sound always plays through both my speakers and my headphones regardless except the sound through the headphones is crackly. It is also a bit quiet even though everything is maxed. I've managed to improve the situation to a point where the sound out of my speakers is perfect but I have none at all from my headphones. I do have two connectors listed in the sound settings but regardless of which one is selected it always plays through the speakers. I think this might have something to do with the fact that my speakers are plugging into the front of my computer, typically the headphone jack, and my headphones are plugging into the back but when I try disconnecting the speakers from the front there is still no sound from the headphones. I fixed the speaker sound by going through the sound settings and making sure they were all set to 100% then rebooting. Things I have tried: Maxing everything and unmuting everything in alsamixer Uninstalling pulseaudio Making gstreamer use only alsa via gstreamer-properties. This worked with the sound test button including independent sound between headphones and speakers but when I reset the computer it no longer worked. So I tried setting it manually in gconf-editor which didn't work either. Reinstalling alsa and pulseaudio Setting the model in /etc/modprobe.d/alsa-base.conf to 6stack and 6stack-dig neither worked. Upgrading to 12.10 Here's some command output to help you diagnose my problem. aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: AD198x Digital [AD198x Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 2: AD198x Headphone [AD198x Headphone] Subdevices: 1/1 Subdevice #0: subdevice #0 sudo lshw -C sound *-multimedia description: Audio device product: 82801JI (ICH10 Family) HD Audio Controller vendor: Intel Corporation physical id: 1b bus info: pci@0000:00:1b.0 version: 00 width: 64 bits clock: 33MHz capabilities: pm msi pciexpress bus_master cap_list configuration: driver=snd_hda_intel latency=0 resources: irq:70 memory:f7ff8000-f7ffbfff cat /proc/asound/card*/codec* | grep "Codec" Codec: Analog Devices AD1989B cat /etc/modprobe.d/alsa-base.conf # autoloader aliases install sound-slot-0 /sbin/modprobe snd-card-0 install sound-slot-1 /sbin/modprobe snd-card-1 install sound-slot-2 /sbin/modprobe snd-card-2 install sound-slot-3 /sbin/modprobe snd-card-3 install sound-slot-4 /sbin/modprobe snd-card-4 install sound-slot-5 /sbin/modprobe snd-card-5 install sound-slot-6 /sbin/modprobe snd-card-6 install sound-slot-7 /sbin/modprobe snd-card-7 # Cause optional modules to be loaded above generic modules install snd /sbin/modprobe --ignore-install snd $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-ioctl32 ; /sbin/modprobe --quiet --use-blacklist snd-seq ; } # # Workaround at bug #499695 (reverted in Ubuntu see LP #319505) install snd-pcm /sbin/modprobe --ignore-install snd-pcm $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-pcm-oss ; : ; } install snd-mixer /sbin/modprobe --ignore-install snd-mixer $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-mixer-oss ; : ; } install snd-seq /sbin/modprobe --ignore-install snd-seq $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; /sbin/modprobe --quiet --use-blacklist snd-seq-oss ; : ; } # install snd-rawmidi /sbin/modprobe --ignore-install snd-rawmidi $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; : ; } # Cause optional modules to be loaded above sound card driver modules install snd-emu10k1 /sbin/modprobe --ignore-install snd-emu10k1 $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-emu10k1-synth ; } install snd-via82xx /sbin/modprobe --ignore-install snd-via82xx $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq ; } # Load saa7134-alsa instead of saa7134 (which gets dragged in by it anyway) install saa7134 /sbin/modprobe --ignore-install saa7134 $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist saa7134-alsa ; : ; } # Prevent abnormal drivers from grabbing index 0 options bt87x index=-2 options cx88_alsa index=-2 options saa7134-alsa index=-2 options snd-atiixp-modem index=-2 options snd-intel8x0m index=-2 options snd-via82xx-modem index=-2 options snd-usb-audio index=-2 options snd-usb-caiaq index=-2 options snd-usb-ua101 index=-2 options snd-usb-us122l index=-2 options snd-usb-usx2y index=-2 # Ubuntu #62691, enable MPU for snd-cmipci options snd-cmipci mpu_port=0x330 fm_port=0x388 # Keep snd-pcsp from being loaded as first soundcard options snd-pcsp index=-2 # Keep snd-usb-audio from beeing loaded as first soundcard options snd-usb-audio index=-2 options snd-hda-intel model=6stack

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  • SBT run differences between scala and java?

    - by Eric Cartner
    I'm trying to follow the log4j2 configuration tutorials in a SBT 0.12.1 project. Here is my build.sbt: name := "Logging Test" version := "0.0" scalaVersion := "2.9.2" libraryDependencies ++= Seq( "org.apache.logging.log4j" % "log4j-api" % "2.0-beta3", "org.apache.logging.log4j" % "log4j-core" % "2.0-beta3" ) When I run the main() defined in src/main/scala/logtest/Foo.scala: package logtest import org.apache.logging.log4j.{Logger, LogManager} object Foo { private val logger = LogManager.getLogger(getClass()) def main(args: Array[String]) { logger.trace("Entering application.") val bar = new Bar() if (!bar.doIt()) logger.error("Didn't do it.") logger.trace("Exiting application.") } } I get the output I was expecting given that src/main/resources/log4j2.xml sets the root logging level to trace: [info] Running logtest.Foo 08:39:55.627 [run-main] TRACE logtest.Foo$ - Entering application. 08:39:55.630 [run-main] TRACE logtest.Bar - entry 08:39:55.630 [run-main] ERROR logtest.Bar - Did it again! 08:39:55.630 [run-main] TRACE logtest.Bar - exit with (false) 08:39:55.630 [run-main] ERROR logtest.Foo$ - Didn't do it. 08:39:55.630 [run-main] TRACE logtest.Foo$ - Exiting application. However, when I run the main() defined in src/main/java/logtest/LoggerTest.java: package logtest; import org.apache.logging.log4j.Logger; import org.apache.logging.log4j.LogManager; public class LoggerTest { private static Logger logger = LogManager.getLogger(LoggerTest.class.getName()); public static void main(String[] args) { logger.trace("Entering application."); Bar bar = new Bar(); if (!bar.doIt()) logger.error("Didn't do it."); logger.trace("Exiting application."); } } I get the output: [info] Running logtest.LoggerTest ERROR StatusLogger Unable to locate a logging implementation, using SimpleLogger ERROR Bar Did it again! ERROR LoggerTest Didn't do it. From what I can tell, ERROR StatusLogger Unable to ... is usually a sign that log4j-core is not on my classpath. The lack of TRACE messages seems to indicate that my log4j2.xml settings aren't on the classpath either. Why should there be any difference in classpath if I'm running Foo.main versus LoggerTest.main? Or is there something else causing this behavior? Update I used SBT Assembly to build a fat jar of this project and specified logtest.LoggerTest to be the main class. Running it from the command line produced correct results: Eric-Cartners-iMac:target ecartner$ java -jar "Logging Test-assembly-0.0.jar" 10:52:23.220 [main] TRACE logtest.LoggerTest - Entering application. 10:52:23.221 [main] TRACE logtest.Bar - entry 10:52:23.221 [main] ERROR logtest.Bar - Did it again! 10:52:23.221 [main] TRACE logtest.Bar - exit with (false) 10:52:23.221 [main] ERROR logtest.LoggerTest - Didn't do it. 10:52:23.221 [main] TRACE logtest.LoggerTest - Exiting application.

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  • With NHibernate, how can I create an INHibernateProxy?

    - by Eric
    After lots of reading about serialization, I've decided to try to create DTOs. After more reading, I decided to use AutoMapper. What I would like to do is transform the parent (easy enough) and transform the entity properties if they've been initialized, which I've done with ValueResolvers like below (I may try to make it generic once I get it fully working). This part works. public class OrderItemResolver : ValueResolver<Order, OrderItem> { protected override OrderItem ResolveCore(Order source) { // could also use NHibernateUtil.IsInitialized(source.OrderItem) if (source.OrderItem is NHibernate.Proxy.INHibernateProxy) return null; else return source.OrderItem; } } } When I transform the DTO back to an entity, for the entities that weren't initialized, I want to create a proxy so that if the entity wants to access it, it can. However, I can't figure out how to create a proxy. I'm using Castle if that's relevant. I've tried a bunch of things with no luck. The below code is a mess, mainly because I've been trying things at random without knowing what I should be doing. Anybody have any suggestions? public class OrderItemDTOResolver : ValueResolver<OrderDTO, OrderItem> { protected override OrderItem ResolveCore(OrderDTO source) { if (source.OrderItem == null) { //OrderItem OrderItem = new ProxyGenerator().CreateClassProxy<OrderItem>(); // Castle.Core.Interceptor. //OrderItem OrderItem = new ProxyGenerator().CreateClassProxy<OrderItem>(); //OrderItem.Id = source.OrderItemId; //OrderItem OrderItem = new OrderItem(); //var proxy = new OrderItem() as INHibernateProxy; //var proxy = OrderItem as INHibernateProxy; //return (OrderItem)proxy.HibernateLazyInitializer //ILazyInitializer proxy = new LazyInitializer("OrderItem", OrderItem, source.OrderItemId, null, null, null, null); //return (OrderItem)proxy; //return (OrderItem)proxy.HibernateLazyInitializer.GetImplementation(); //return OrderItem; IProxyTargetAccessor proxy = new Castle.Core.Interceptor. var initializer = new LazyInitializer("OrderItem", typeof(OrderItem), source.OrderItemId, null, null, null, null); //var proxyFactory = new SerializableProxyFactory{Interfaces = Interfaces, TargetSource = initializer, ProxyTargetType = IsClassProxy}; //proxyFactory.AddAdvice(initializer); //object proxyInstance = proxyFactory.GetProxy(); //return (INHibernateProxy) proxyInstance; return null; //OrderItem.Id = source.OrderItemId; //return OrderItem; } else return OrderItemDTO.Unmap(source.OrderItem); } } Thanks, Eric

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  • compressed archive with quick access to individual file

    - by eric.frederich
    I need to come up with a file format for new application I am writing. This file will need to hold a bunch other text files which are mostly text but can be other formats as well. Naturally, a compressed tar file seems to fit the bill. The problem is that I want to be able to retrieve some data from the file very quickly and getting just a particular file from a tar.gz file seems to take longer than it should. I am assumeing that this is because it has to decompress the entire file even though I just want one. When I have just a regular uncompressed tar file I can get that data real quick. Lets say the file I need quickly is called data.dat For example the command... tar -x data.dat -zf myfile.tar.gz ... is what takes a lot longer than I'd like. MP3 files have id3 data and jpeg files have exif data that can be read in quickly without opening the entire file. I would like my data.dat file to be available in a similar way. I was thinking that I could leave it uncompressed and seperate from the rest of the files in myfile.tar.gz I could then create a tar file of data.dat and myfile.tar.gz and then hopefully that data would be able to be retrieved faster because it is at the head of outer tar file and is uncompressed. Does this sound right?... putting a compressed tar inside of a tar file? Basically, my need is to have an archive type of file with quick access to one particular file. Tar does this just fine, but I'd also like to have that data compressed and as soon as I do that, I no longer have quick access. Are there other archive formats that will give me that quick access I need? As a side note, this application will be written in Python. If the solution calls for a re-invention of the wheel with my own binary format I am familiar with C and would have no problem writing the Python module in C. Idealy I'd just use tar, dd, cat, gzip, etc though. Thanks, ~Eric

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  • Building a Mafia&hellip;TechFest Style

    - by David Hoerster
    It’s been a few months since I last blogged (not that I blog much to begin with), but things have been busy.  We all have a lot going on in our lives, but I’ve had one item that has taken up a surprising amount of time – Pittsburgh TechFest 2012.  After the event, I went through some minutes of the first meetings for TechFest, and I started to think about how it all came together.  I think what inspired me the most about TechFest was how people from various technical communities were able to come together and build and promote a common event.  As a result, I wanted to blog about this to show that people from different communities can work together to build something that benefits all communities.  (Hopefully I've got all my facts straight.)  TechFest started as an idea Eric Kepes and myself had when we were planning our next Pittsburgh Code Camp, probably in the summer of 2011.  Our Spring 2011 Code Camp was a little different because we had a great infusion of some folks from the Pittsburgh Agile group (especially with a few speakers from LeanDog).  The line-up was great, but we felt our audience wasn’t as broad as it should have been.  We thought it would be great to somehow attract other user groups around town and have a big, polyglot conference. We started contacting leaders from Pittsburgh’s various user groups.  Eric and I split up the ones that we knew about, and we just started making contacts.  Most of the people we started contacting never heard of us, nor we them.  But we all had one thing in common – we ran user groups who’s primary goal is educating our members to make them better at what they do. Amazingly, and I say this because I wasn’t sure what to expect, we started getting some interest from the various leaders.  One leader, Greg Akins, is, in my opinion, Pittsburgh’s poster boy for the polyglot programmer.  He’s helped us in the past with .NET Code Camps, is a Java developer (and leader in Pittsburgh’s Java User Group), works with Ruby and I’m sure a handful of other languages.  He helped make some e-introductions to other user group leaders, and the whole thing just started to snowball. Once we realized we had enough interest with the user group leaders, we decided to not have a Fall Code Camp and instead focus on this new entity. Flash-forward to October of 2011.  I set up a meeting, with the help of Jeremy Jarrell (Pittsburgh Agile leader) to hold a meeting with the leaders of many of Pittsburgh technical user groups.  We had representatives from 12 technical user groups (Python, JavaScript, Clojure, Ruby, PittAgile, jQuery, PHP, Perl, SQL, .NET, Java and PowerShell) – 14 people.  We likened it to a scene from a Godfather movie where the heads of all the families come together to make some deal.  As a result, the name “TechFest Mafia” was born and kind of stuck. Over the next 7 months or so, we had our starts and stops.  There were moments where I thought this event would not happen either because we wouldn’t have the right mix of topics (was I off there!), or enough people register (OK, I was wrong there, too!) or find an appropriate venue (hmm…wrong there, too) or find enough sponsors to help support the event (wow…not doing so well).  Overall, everything fell into place with a lot of hard work from Eric, Jen, Greg, Jeremy, Sean, Nicholas, Gina and probably a few others that I’m forgetting.  We also had a bit of luck, too.  But in the end, the passion that we had to put together an event that was really about making ourselves better at what we do really paid off. I’ve never been more excited about a project coming together than I have been with Pittsburgh TechFest 2012.  From the moment the first person arrived at the event to the final minutes of my closing remarks (where I almost lost my voice – I ended up being diagnosed with bronchitis the next day!), it was an awesome event.  I’m glad to have been part of bringing something like this to Pittsburgh…and I’m looking forward to Pittsburgh TechFest 2013.  See you there!

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  • How to disable Bluetooth auto-answer on Windows 7?

    - by MarkR
    When my BlackBerry 9630 is connected via Bluetooth to my Dell desktop running Windows 7 x64 there are a bunch of Bluetooth services enabled on Windows Advanced Audio BB Bypass service BB Desktop Service Dial-up Networking Headset Audio Gateway Remote Control Remotely Controlled Device When a call arrives on the 9630 Windows immediately answer the call. This is annoying when windows answers the phone before I even hear an audible ring and the hapless caller is saying, "hello. hello" on top of music I have playing. Does anyone know how to tell Windows not to answer the call? I want to be able to don my headset, switch audio to my headset, THEN manually answer the call.

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  • v4l - capture and watch at the same time

    - by John Barrett
    Capturing v4l and line-in audio using mencoder works very well, but I would like to record real-time gameplay video from consoles plugged into the video card. I've used xawtv for this (Works quite well, can preview and record in real time), but when I enable any deinterlacing or aspect ration options the video fails to record. I have to record raw and re-encode the video with the appropriate filters later to get something workable. Other things I have tried: tvtime with xvidcap and jack audio capture - xvidcap drops frames and muxing the audio is impossible as it will go out of sync (I have not found muxer options that work to force a correct frame rate) mencoder capture to file, attempt to pipe tail of file to mplayer... mencoder works great, piping the file is far too heavy to attempt gameplay. Soooo, v4l capture and preview simultaneously, recommendations?

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  • graphics cards with no HDMI output

    - by Noam Gal
    I am currently looking at Gainward GTX260 896MB GS GLH, but I've seen it also on the x275 version - the card doesn't have an HDMI output, only two dvi and one tv out (s-video?). They claim they support HDMI using a dvi-HDMI converter. Will I get a true high definition quality on my TV (assuming it supports it) like that? Or is it not as good, and I should stick to cards that have an HDMI output (ATI), or pay way too much for x295? What about connecting the audio? The x260 comes with an internal spdif cable - does that mean I can connect my soundcard to my graphics card, and have the audio come out through the dvi, and into the HDMI cable? Or am I mixing it all up here, and I have to somehow connect the sound to the TV using a seperate cable (Hoping it has a seperate audio-in for the HDMI channel)?

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  • Enabling Surround sound on a Realtek ALC892 via SPDIF, on Windows 7

    - by Alex
    I have a problem with my ALC892, on an ASRock mainboard (ASRock 890FX Deluxe4). I get only stereo sounds if I use SPDIF connection, in general. My amp shows that is getting surround sound only when I use the Test feature of Windows 7. This test feature allows to know which formats are supported by the audio chip. The tests render correctly both Dolby Digital and DTS. You can find this test under Sounds, Playback Devices, Select Digital Audio, then "Properties". I am using Windows 7 x64, with the latest drivers from the official Realtek website. I also tested other driver versions, both from the Realtek website and from the ASRock one, but had no luck. Thanks for the help. Some specs: CPU: AMD Phenom II X4 965 MOBO: ASRock 890FX Deluxe4 (with onboard Realtek ALC892) Audio amp: Onkyo R-380 (works fine with other sources like PS3 and Xbox 360)

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  • Use external speakers with laptop hooked to separate monitor?

    - by lhan16
    I have a laptop with a set of external speakers hooked up to it on my computer desk. The speakers use the standard 3.5mm audio (headphones) jack. The speakers work fine, but I've recently added a separate monitor to my laptop via HDMI. With the monitor hooked up to my laptop and the speakers still hooked up to the laptop, sound will only come out of the built-in monitor speakers. When I look at my audio settings, there are three different "audio playback devices" showing up, but only the built-in monitor speakers make noise when I click "test" (and I hear nothing when I set any of the other devices as the default. Does anyone know how I can still use my external speakers when using a separate monitor with my laptop? I'm hoping there is a solution that doesn't require the laptop to be open or closed, because I use both scenarios. I came across this post, but it doesn't look like they had much luck.

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  • Android: Use XML Layout for List Cell rather than Java Code Layout (Widgets)

    - by Stephen Finucane
    Hi, I'm in the process of making a music app and I'm currently working on the library functionality. I'm having some problems, however, in working with a list view (In particular, the cells). I'm trying to move from a simple textview layout in each cell that's created within java to one that uses an XML file for layout (Hence keeping the Java file mostly semantic) This is my original code for the cell layout: public View getView(int position, View convertView, ViewGroup parent) { String id = null; TextView tv = new TextView(mContext.getApplicationContext()); if (convertView == null) { music_column_index = musiccursor.getColumnIndexOrThrow(MediaStore.Audio.Media.TITLE); musiccursor.moveToPosition(position); id = musiccursor.getString(music_column_index); music_column_index = musiccursor.getColumnIndexOrThrow(MediaStore.Audio.Media.DISPLAY_NAME); musiccursor.moveToPosition(position); id += "\n" + musiccursor.getString(music_column_index); music_column_index = musiccursor.getColumnIndexOrThrow(MediaStore.Audio.Albums.ALBUM); musiccursor.moveToPosition(position); id += "\n" + musiccursor.getString(music_column_index); tv.setText(id); } else tv = (TextView) convertView; return tv; } And my new version: public View getView(int position, View convertView, ViewGroup parent) { View cellLayout = findViewById(R.id.albums_list_cell); ImageView album_art = (ImageView) findViewById(R.id.album_cover); TextView album_title = (TextView) findViewById(R.id.album_title); TextView artist_title = (TextView) findViewById(R.id.artist_title); if (convertView == null) { music_column_index = musiccursor.getColumnIndexOrThrow(MediaStore.Audio.Albums.ALBUM); musiccursor.moveToPosition(position); album_title.setText(musiccursor.getString(music_column_index)); //music_column_index = musiccursor.getColumnIndexOrThrow(MediaStore.Audio.Media.DISPLAY_NAME); //musiccursor.moveToPosition(position); music_column_index = musiccursor.getColumnIndexOrThrow(MediaStore.Audio.Media.TITLE); musiccursor.moveToPosition(position); artist_title.setText(musiccursor.getString(music_column_index)); } else{ cellLayout = (TextView) convertView; } return cellLayout; } The initialisation (done in the on create file): musiclist = (ListView) findViewById(R.id.PhoneMusicList); musiclist.setAdapter(new MusicAdapter(this)); musiclist.setOnItemClickListener(musicgridlistener); And the respective XML files: (main) <?xml version="1.0" encoding="utf-8"?> <LinearLayout xmlns:android="http://schemas.android.com/apk/res/android" android:orientation="vertical" android:layout_width="fill_parent" android:layout_height="fill_parent"> <ListView android:id="@+id/PhoneMusicList" android:layout_width="fill_parent" android:layout_height="fill_parent" /> <TextView android:id="@android:id/empty" android:layout_width="wrap_content" android:layout_height="0dip" android:layout_weight="1.0" android:text="@string/no_list_data" /> </LinearLayout> (albums_list_cell) <?xml version="1.0" encoding="utf-8"?> <RelativeLayout xmlns:android="http://schemas.android.com/apk/res/android" android:id="@+id/albums_list_cell" android:layout_width="wrap_content" android:layout_height="wrap_content"> <ImageView android:id="@+id/album_cover" android:layout_alignParentLeft="true" android:layout_alignParentTop="true" android:layout_width="50dip" android:layout_height="50dip" /> <TextView android:id="@+id/album_title" android:layout_toRightOf="@+id/album_cover" android:layout_alignParentTop="true" android:layout_width="wrap_content" android:layout_height="wrap_content" /> <TextView android:id="@+id/artist_title" android:layout_toRightOf="@+id/album_cover" android:layout_below="@+id/album_title" android:layout_width="wrap_content" android:layout_height="15dip" /> </RelativeLayout> In theory (based on the tiny bit of Android I've done so far) this should work..it doesn't though. Logcat gives me a null pointer exception at line 96 of the faulty code, which is the album_title.setText line. It could be a problem with my casting but Google tells me this is ok :D Thanks for any help and let me know if you need more info!

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  • XPP-32 over W7-64 on music production laptop

    - by quarlo
    I need to upgrade my laptop and need high performance for music production (recording and mixing). My audio interface manufacturer seems to be unable to successfully convert their drivers to 64-bit. I do not trust a virtual machine to handle real-time audio recording at low enough latency so ... I would like to install XP Pro 32-bit on a separate partition and dual boot since most of the machines that can handle this application now ship with Windows 7 64-bit flavors. I'd like to transit to 64-bit over time assuming M-Audio does eventually get a handle on 64-bit drivers, but really need to ensure that I can stay at 32-bit for now. Does anyone have any experience with this or something similar?

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