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  • rotating spheres

    - by Dave
    I want to continuously rotate 2 spheres, however the rotation does not seem to work. Here is my code: float angle = 0.0f; void light(){ glEnable(GL_LIGHTING); glEnable(GL_LIGHT0); glEnable(GL_LIGHT1); // Create light components GLfloat positionlight1[] = { 9.0, 5.0, 1.0, 0.0 }; GLfloat positionlight2[] = {0.2,2.5,1.3,0.0}; GLfloat light_ambient1[] = { 0.0, 0.0, 1.0, 1.0}; GLfloat light_diffuse[] = { 1.0, 1.0, 1.0, 1.0 }; glLightfv(GL_LIGHT0, GL_AMBIENT, light_ambient1); glLightfv(GL_LIGHT1, GL_DIFFUSE, light_diffuse); glLightfv(GL_LIGHT0, GL_POSITION, positionlight1); glLightfv(GL_LIGHT1, GL_POSITION, positionlight2); } void changeSize(int w, int h) { if (h==0) // Prevent A Divide By Zero By { h=1; // Making Height Equal One } glMatrixMode(GL_PROJECTION); // Select The Projection Matrix glLoadIdentity(); // Reset The Projection Matrix glViewport(0,0,w,h);// Reset The Current Viewport // Calculate The Aspect Ratio Of The Window gluPerspective(45.0f,(GLfloat)w/(GLfloat)h,0.1f,100.0f); glMatrixMode(GL_MODELVIEW); // Select The Modelview Matrix // Reset The Modelview Matrix } void renderScene(void) { glClear(GL_COLOR_BUFFER_BIT | GL_DEPTH_BUFFER_BIT); glPushMatrix(); //set where to start the current object glTranslatef(0.0,1.2,-6); glRotatef(angle,0,1.2,-6); glutSolidSphere(1,50,50); glPopMatrix(); //end the current object transformations glPushMatrix(); //set where to start the current object glTranslatef(0.0,-2,-6); glRotatef(angle,0,-2,-6); glutSolidSphere(0.5,50,50); glPopMatrix(); //end the current object transformations angle=+0.1; glutSwapBuffers(); } int main(int argc, char **argv) { // init GLUT and create window glutInit(&argc, argv); glutInitDisplayMode(GLUT_DEPTH | GLUT_DOUBLE | GLUT_RGBA); glutInitWindowPosition(100,100); glutInitWindowSize(500,500); glutCreateWindow("Hello World"); // register callbacks light(); glutDisplayFunc(renderScene); glutReshapeFunc(changeSize); glutIdleFunc(renderScene); // enter GLUT event processing loop glutMainLoop(); return 1; } Graphicstest::Graphicstest(void) { } In the renderscene where i draw,translate and rotate my 2 spheres. It does not seem to rotate the spheres continuously. What am i doing wrong?

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  • Multi-threaded JOGL Problem

    - by moeabdol
    I'm writing a simple OpenGL application in Java that implements the Monte Carlo method for estimating the value of PI. The method is pretty easy. Simply, you draw a circle inside a unit square and then plot random points over the scene. Now, for each point that is inside the circle you increment the counter for in points. After determining for all the random points wither they are inside the circle or not you divide the number of in points over the total number of points you have plotted all multiplied by 4 to get an estimation of PI. It goes something like this PI = (inPoints / totalPoints) * 4. This is because mathematically the ratio of a circle's area to a square's area is PI/4, so when we multiply it by 4 we get PI. My problem doesn't lie in the algorithm itself; however, I'm having problems trying to plot the points as they are being generated instead of just plotting everything at once when the program finishes executing. I want to give the application a sense of real-time display where the user would see the points as they are being plotted. I'm a beginner at OpenGL and I'm pretty sure there is a multi-threading feature built into it. Non the less, I tried to manually create my own thread. Each worker thread plots one point at a time. Following is the psudo-code: /* this part of the code exists in display() method in MyCanvas.java which extends GLCanvas and implements GLEventListener */ // main loop for(int i = 0; i < number_of_points; i++){ RandomGenerator random = new RandomGenerator(); float x = random.nextFloat(); float y = random.nextFloat(); Thread pointThread = new Thread(new PointThread(x, y)); } gl.glFlush(); /* this part of the code exists in run() method in PointThread.java which implements Runnable */ void run(){ try{ gl.glPushMatrix(); gl.glBegin(GL2.GL_POINTS); if(pointIsIn) gl.glColor3f(1.0f, 0.0f, 0.0f); // red point else gl.glColor3f(0.0f, 0.0f, 1.0f); // blue point gl.glVertex3f(x, y, 0.0f); // coordinates gl.glEnd(); gl.glPopMatrix(); }catch(Exception e){ } } I'm not sure if my approach to solving this issue is correct. I hope you guys can help me out. Thanks.

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  • pfSense: How to route traffic out the WAN port?

    - by Ian Boyd
    Expert version i want to create a route in pfSense that will send traffic out the physical WAN port, not the PPPoE WAN port. i want to talk to talk to the web-server on my DSL modem, but it doesn't see packets wrapped in a PPPoE header. Long version My pfSense router is responsible for setting up the PPPoE connection over DSL to my ISP. When a machine on the LAN wants to sent packets to the internet, the default route sends packets out over the PPPoE connection. Those packets, wrapped in a PPPoE header, are sent on the ethernet cable to my DSL modem. From there they are sent the ISP, and the internet at large. i want a way to send a packet out the WAN port itself - not the PPPoE WAN port. My modem is sitting out there, with a http interface where i can monitor connection speed signal-to-noise ratio bandwidth connection time Whenever i try to set a route for destination of 192.168.2.1 (the IP that the modem will listen to for HTTP requests) to go out the WAN port, they instead end up going out the PPPoE port. The difference being that they're wrapped in a PPPoE protocol packet, and the modem isn't being sent the packet, it's being delivered to the ISP. Given that pfSense has no ability to direct traffic out the physical WAN port: how can i direct traffic out the physical WAN port on pfSense?

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  • High Sqlservr.exe Memory Usage

    - by user18576
    I have a problem with sqlservr.exe (version 2008). It use a more memory. I checked on windows taskbar manager, sqlservr.exe usage ( Mem usage - 8GB Ram). I dont know how can I fix it.Got the following metrics of the server using Perfmon: SQLServer:Buffer Manager Buffer cache hit ratio 13 SQLServer:Buffer Manager Page lookups/sec 46026128096 SQLServer:Buffer Manager Free pages 129295 SQLServer:Buffer Manager Total pages 997309 SQLServer:Buffer Manager Target pages 1053560 SQLServer:Buffer Manager Database pages 484117 SQLServer:Buffer Manager Reserved pages 0 SQLServer:Buffer Manager Stolen pages 383897 SQLServer:Buffer Manager Lazy writes/sec 384369 SQLServer:Buffer Manager Readahead pages/sec 69315446 SQLServer:Buffer Manager Page reads/sec 71280353 SQLServer:Buffer Manager Page writes/sec 12408371 SQLServer:Buffer Manager Checkpoint pages/sec 7053801 SQLServer:Buffer Manager Page life expectancy 735262 SQLServer:General Statistics Active Temp Tables 161 SQLServer:General Statistics Temp Tables Creation Rate 3131845 SQLServer:General Statistics Logins/sec 2336011 SQLServer:General Statistics Logouts/sec 2335984 SQLServer:General Statistics User Connections 27 SQLServer:General Statistics Transactions 0 SQLServer:Access Methods Full Scans/sec 34422821 SQLServer:Access Methods Range Scans/sec 2027247756 SQLServer:Access Methods Workfiles Created/sec 49771600 SQLServer:Access Methods Worktables Created/sec 28205828 SQLServer:Access Methods Index Searches/sec 4890715219 SQLServer:Access Methods FreeSpace Scans/sec 21178928 SQLServer:Access Methods FreeSpace Page Fetches/sec 21226653 SQLServer:Access Methods Pages Allocated/sec 41483279 SQLServer:Access Methods Extents Allocated/sec 4743504 SQLServer:Access Methods Extent Deallocations/sec 4806606 SQLServer:Access Methods Page Deallocations/sec 41419137 SQLServer:Access Methods Page Splits/sec 23834799 SQLServer:Memory Manager SQL Cache Memory (KB) 29160 SQLServer:Memory Manager Target Server Memory (KB) 8428480 SQLServer:Memory Manager Total Server Memory (KB) 7978472 Some body could help me please.And I really want to know the cause for the above.

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  • Is it possible to shrink the size of an HP Smart Array logical drive?

    - by ewwhite
    I know extension is quite possible using the hpacucli utility, but is there an easy way to reduce the size of an existing logical drive (not array)? The controller is a P410i in a ProLiant DL360 G6 server. I'd like to reduce logicaldrive 1 from 72GB to 40GB. => ctrl all show config detail Smart Array P410i in Slot 0 (Embedded) Bus Interface: PCI Slot: 0 Serial Number: 5001438006FD9A50 Cache Serial Number: PAAVP9VYFB8Y RAID 6 (ADG) Status: Disabled Controller Status: OK Chassis Slot: Hardware Revision: Rev C Firmware Version: 3.66 Rebuild Priority: Medium Expand Priority: Medium Surface Scan Delay: 3 secs Surface Scan Mode: Idle Queue Depth: Automatic Monitor and Performance Delay: 60 min Elevator Sort: Enabled Degraded Performance Optimization: Disabled Inconsistency Repair Policy: Disabled Wait for Cache Room: Disabled Surface Analysis Inconsistency Notification: Disabled Post Prompt Timeout: 15 secs Cache Board Present: True Cache Status: OK Accelerator Ratio: 25% Read / 75% Write Drive Write Cache: Enabled Total Cache Size: 512 MB No-Battery Write Cache: Disabled Cache Backup Power Source: Batteries Battery/Capacitor Count: 1 Battery/Capacitor Status: OK SATA NCQ Supported: True Array: A Interface Type: SAS Unused Space: 412476 MB Status: OK Logical Drive: 1 Size: 72.0 GB Fault Tolerance: RAID 1+0 Heads: 255 Sectors Per Track: 32 Cylinders: 18504 Strip Size: 256 KB Status: OK Array Accelerator: Enabled Unique Identifier: 600508B1001C132E4BBDFAA6DAD13DA3 Disk Name: /dev/cciss/c0d0 Mount Points: /boot 196 MB, / 12.0 GB, /usr 8.0 GB, /var 4.0 GB, /tmp 2.0 GB OS Status: LOCKED Logical Drive Label: AE438D6A5001438006FD9A50BE0A Mirror Group 0: physicaldrive 1I:1:1 (port 1I:box 1:bay 1, SAS, 146 GB, OK) physicaldrive 1I:1:2 (port 1I:box 1:bay 2, SAS, 146 GB, OK) Mirror Group 1: physicaldrive 1I:1:3 (port 1I:box 1:bay 3, SAS, 146 GB, OK) physicaldrive 1I:1:4 (port 1I:box 1:bay 4, SAS, 146 GB, OK) SEP (Vendor ID PMCSIERA, Model SRC 8x6G) 250 Device Number: 250 Firmware Version: RevC WWID: 5001438006FD9A5F Vendor ID: PMCSIERA Model: SRC 8x6G

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  • What is the best / latest software to convert DVD recorder disc to YouTube?

    - by Jian Lin
    I recorded some Wii gameplay to a DVD-R using a DVD recorder, and tried using Handbrake to convert it to YouTube compatible format. Basically, it needs resizing to 16:9, and selecting 00:03 to 3:10 of the chapter (out of the 7 minutes). It seems Handbrake can't do it for 480p... and if using AutoGK and then VirtualDub to resize and do video time selection, it feels so old school (autogk not updated since 2005)... Windows Movie Maker seems like 15fps video quality... Is there a way to do it nicely? Details if using Handbrake: I can't find a way to convert a DVD recorded video which is 720 x 480 (Wii Gameplay in 16 : 9 ratio) into 852 x 480... It seems that the only possible resize is to shrink but not to expand? It is fine to make it 720 x 406... except YouTube won't be able to show it as 480p... If it can be made 852 x 480, then YouTube can at least show it as 480p (besides 360p and 240p)... is there any way to do it? thanks.

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  • Mplayer no sound when playing some movies

    - by Ivan Peevski
    Ok, that's a bit of a strange problem, that somehow crept into my system. It used to work fine. Here is the problem as far as I can identify it. When I try to play certain video files with mplayer, there is no sound. As far as I can tell, it is only an issue with ac3 and dts sound tracks (using the ffmpeg decoder). Mplayer says: ========================================================================== Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders AUDIO: 48000 Hz, 6 ch, s16le, 1536.0 kbit/33.33% (ratio: 192000->576000) Selected audio codec: [ffdca] afm: ffmpeg (FFmpeg DTS) ========================================================================== [AO_ALSA] Playback open error: Device or resource busy Failed to initialize audio driver 'alsa' Could not open/initialize audio device -> no sound. Audio: no sound (similar with ac3 sound, but using the ffac3 audio codec). Trying different audio output (-ao oss/pcm/sdl) doesn't fix the problem. The strange thing is that if I play these files directly with ffplay, they work fine. mplayer sound with mp3/ogg is fine My alsa configuration is standard (no /etc/asound.conf or ~/.asound*) OS: Linux Gentoo Mplayer: 1.0_rc4_p20100213 (SVN-r30554-4.3.4) FFMpeg: 0.5_p20601-r1 (SVN-r20601) Any other information I can provide?

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  • Using NginX and Apache alongside for both static and dynamic files

    - by faridv
    Background: I've searched a lot and found these useful threads about using of Apache or NginX for static or dynamic files. But they are old (mostly about 1 or 2 years ago) and I think both webservers, specifically Nginx has had important changes in performance and usage. So I think take another look on these issue cannot be that bad. Nginx (for static files) and Apache (for dynamic content)? nginx better than apache for dynamic content? [closed] Apache or NGINX for PHP? Nginx as reverse proxy to Apache with only dynamic content? My question: I have a PHP web application with lots of dynamic files and lots of static contents (videos, images etc.) and it's currently running on a CentOS 6 server and Apache 2.2 since 2 months ago. In past few days, number of our site visitors have gained so fast and I just thought if this number continues to increase with current ratio, we need to change many things (web server, application, etc.) to prevent failures. Because of hardware limitations that we are facing, I thought that it's best for us to start with web server. Should I start with something else? Should I try to increase performance of my PHP application and forget about web server for now? (even if gonna take a long time!) Because of huge usage of .htaccess files (for redirection, rewrites, etc.), I think it's gonna be painful to migrate to NginX as default web server or maybe only for dynamic files. Does this mean that I can't even use Nginx as reverse proxy? I'm not sure latest stable version of NginX and PHP-FPM have a better performance over my current Apache and my limitations (too many things) won't let me to give it a try. Which one is doing better currently? What will I lose by migrating to Nginx? To make it short, what should I do?

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  • Safari's location bar (auto-suggest and web search)

    - by Lri
    Auto-suggest don't seem to work for queries with spaces. Am I missing something? If you select an item from the suggestion list that was matched by its title, the title is filled in before the address. Can you change it to work like in other browsers? SMRT disables searching by title completely. Can you combine Top Hit, History and Bookmarks into a single section? The preferences starting with DebugSafari4 don't work anymore. (Like DebugSafari4IncludeFancyURLCompletionList.) Can you direct unresolved addresses to something like google.com/search?q=?&btnI instead of ?.com? Like by changing keyword.URL in Firefox. Can you remove or hide the web search field? In Camino, Cruz and Fluid it can be resized to zero width. You can't circumvent the normal maximum ratio with InputFieldWidthRatio. AddressBarIncludesGoogle doesn't appear do anything in the current version. Are there fixes or workarounds to any of these? I'm lumping these issues together, because they are closely related — a lot of them were introduced when the location bar was redesigned in Safari 5. I'm also hoping to find something like an extension or a plugin that would replace the standard location bar.

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  • Why are FMS logs filled with 'play' event status code 408 for a failed webcast?

    - by Stu Thompson
    Recently we had a live webcast event go horribly wrong. I'm doing the technical post-mortem, with limited information. We know that the hardware encoder (a Digital Rapid Touch Stream Web HDI) was unable to send upstream at a sustained reliable high rate. What we don't know is if the encoder's connection was problematic (Zürich), or that of the streaming server (in Frankfurt). Unfortunately, I've got three different vendors all blaming each other (the CDN who runs the server, the on-site ISP and the on-site encoding team.) In the FMS log files I see a couple of interesting things: Zillions of Status Code 408 on play event entries for clients. Adobe's documentation stats that this "Stream stopped because client disconnected". ("Zillions" would be a ratio of 10 events for every individual IP address.) Several unpublish / (re)publish events per hour for the encoder I'd like to know if all those 408s could tell me with authority that the FMS server was starved for bandwidth, or that the encoding signal was starved (and hence the server was disconnecting clients.) Any clues?

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  • Is basing storage requirements based on IOPS sufficient?

    - by Boden
    The current system in question is running SBS 2003, and is going to be migrated on new hardware to SBS 2008. Currently I'm seeing on average 200-300 disk transfers per second total across all the arrays in the system. The array seeing the bulk of activity is a 6 disk 7200RPM RAID 6 and it struggles to keep up during high traffic times (idle time often only 10-20%; response times peaking 20-50+ ms). Based on some rough calculations this makes sense (avg ~245 IOPS on this array at 70/30 read to write ratio). I'm considering using a much simpler disk configuration using a single RAID 10 array of 10K disks. Using the same parameters for my calculations above, I'm getting 583 average random IOPS / sec. Granted SBS 2008 is not the same beast as 2003, but I'd like to make the assumption that it'll be similar in terms of disk performance, if not better (Exchange 2007 is easier on the disk and there's no ISA server). Am I correct in believing that the proposed system will be sufficient in terms of performance, or am I missing something? I've read so much about recommended disk configurations for various products like Exchange, and they often mention things like dedicating spindles to logs, etc. I understand the reasoning behind this, but if I've got more than enough random I/O overhead, does it really matter? I've always at the very least had separate spindles for the OS, but I could really reduce cost and complexity if I just had a single, good performing array. So as not to make you guys do my job for me, the generic version of this question is: if I have a projected IOPS figure for a new system, is it sufficient to use this value alone to spec the storage, ignoring "best practice" configurations? (given similar technology, not going from DAS to SAN or anything)

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  • RabbitMQ message consumers stop consuming messages

    - by Bruno Thomas
    Hi server fault, Our team is in a spike sprint to choose between ActiveMQ or RabbitMQ. We made 2 little producer/consumer spikes sending an object message with an array of 16 strings, a timestamp, and 2 integers. The spikes are ok on our devs machines (messages are well consumed). Then came the benchs. We first noticed that somtimes, on our machines, when we were sending a lot of messages the consumer was sometimes hanging. It was there, but the messsages were accumulating in the queue. When we went on the bench plateform : cluster of 2 rabbitmq machines 4 cores/3.2Ghz, 4Gb RAM, load balanced by a VIP one to 6 consumers running on the rabbitmq machines, saving the messages in a mysql DB (same type of machine for the DB) 12 producers running on 12 AS machines (tomcat), attacked with jmeter running on another machine. The load is about 600 to 700 http request per second, on the servlets that produces the same load of RabbitMQ messages. We noticed that sometimes, consumers hang (well, they are not blocked, but they dont consume messages anymore). We can see that because each consumer save around 100 msg/sec in database, so when one is stopping consumming, the overall messages saved per seconds in DB fall down with the same ratio (if let say 3 consumers stop, we fall around 600 msg/sec to 300 msg/sec). During that time, the producers are ok, and still produce at the jmeter rate (around 600 msg/sec). The messages are in the queues and taken by the consumers still "alive". We load all the servlets with the producers first, then launch all the consumers one by one, checking if the connexions are ok, then run jmeter. We are sending messages to one direct exchange. All consumers are listening to one persistent queue bounded to the exchange. That point is major for our choice. Have you seen this with rabbitmq, do you have an idea of what is going on ? Thank you for your answers.

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  • How to extract a Vorbis stream from a WAVE file?

    - by H.B.
    I would like to move the Vorbis stream into an ogg container but ffmpeg does not seem to recognize the stream. Even though MPlayer gives this output upon playback: Opening audio decoder: [acm] Win32/ACM decoders Loading codec DLL: 'vorbis.acm' Loaded DLL driver vorbis.acm at 10000000 Warning! ACM codec reports srcsize=0 AUDIO: 44100 Hz, 2 ch, s16le, 128.0 kbit/9.07% (ratio: 16000-176400) Selected audio codec: [vorbisacm] afm: acm (OggVorbis ACM) ffmpeg: ffmpeg -i Source.wav -acodec copy Target.ogg Input #0, wav, from 'Source.wav': Duration: 00:02:15.17, bitrate: 128 kb/s Stream #0.0: Audio: qg[0][0] / 0x6771, 44100 Hz, 2 channels, 128 kb/s [ogg @ 00000000003096C0] Unsupported codec id in stream 0 Output #0, ogg, to 'Target.ogg': Metadata: encoder : Lavf53.6.0 Stream #0.0: Audio: qg[0][0] / 0x6771, 44100 Hz, 2 channels, 128 kb/s Stream mapping: Stream #0.0 -> #0.0 Could not write header for output file #0 (incorrect codec parameters ?) Of course this does not necessarily need to be done via ffmpeg, any method that is workable would be fine... I have cut down one of the files to 512KB: sample.wav (Changed two chunk size fields in the wave header to account for this, the embedded stream is cut "without notice")

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  • Database types for customer analytics

    - by Drewdavid
    I am exploring a paid solution to start providing better embedded, dashboard-style analytics information to our website customers/account holders, but would like to also offer an in-house development option to our team. The more equipped I am with specifics (such as the subject of this question), the better the adoption rate from the team (or so I have found), regardless of the path we choose Would anyone care to summarize a couple of options for a fast and scalable database type through which we would provide the following: • Daily pageviews to a users account pages (users have between 1 and 1000 pages) • Some calculated/compounded metrics (such as conversion rate, i.e. certain page type viewed to contact form thank you page ratio) • We have about 1,500 members (will need room to grow); the number of concurrently logged in users will for the question's sake be 50 I ask because our developer has balked at providing this level of "over time" granularity (i.e. daily) due to the number of space it would take up in a MYSQL database To avoid a downvote I have asked specifically for more than one option, realizing that different people will have different solutions. I will make amendments to my question if so guided by answering parties Thank you for sharing your valued answers :)

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  • ffmpeg: converting an avi to a reduced, shareable flv/mp4...

    - by meder
    I recently followed a guide and recompiled my ffmpeg so x264 is enabled. I used some generic settings to convert my 700 MB avi file to a mp4 file, the result was a 407MB mp4 file. The original avi file's settings: Codec: DX50 Resolution: 704x304 Frame rate: 23.976023 Stream 1 Codec: mpga Type: Audio Channels: 2 Sample rate: 48000 Hz Bitrate 179 kb/s Command I used: ffmpeg -i input.avi -acodec libfaac -ab 128k -ac 2 -vcodec libx264 -vpre hq -crf 22 -threads 0 output.mp4 The settings of the output file (output.mp4): Codec: avc1 Resolution: 704x304 Display resolution: 704x304 Frame rate: 11.988011 Stream 1 Codec: mp4a Type: Audio Channels: 2 Sample Rate: 48000 Hz Bits per sample: 16 Bitrate: 1536 kb/s The quality of the output mp4 is pretty nice, it seems as if it's pretty much the same as the original source. However, I'm trying to reduce the filesize and I'm not really sure whether I should go with an flv format or keep it mp4. The advantage the flv would have obviously is that it would be playable with a flash player ( I have come across some swf players which take a flash parameter to play an flv file ).. but maybe I could use the video element, as I'm only going to be displaying this video privately so I don't have to worry about supporting legacy browsers such as IE. Can someone recommend some settings to specify in order for the filesize to be around ~100-150MB or so? I don't mind a reduction in quality, nor do I mind resizing it - I was going to do it initially but I wasn't sure what the guidelines were ( if any ) for dealing with resolution.. since this video's is 704x304 would it still be ok if I forced it into one that isn't perfectly fit for the aspect ratio? I have no clue about that part. I realize that I could have probably specified 28 instead of 22 for the CRF, I'm not sure if I should do that as opposed to maybe specifying smaller resolution, which might make it smaller as well?

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  • Real benefits of tcp TIME-WAIT and implications in production environment

    - by user64204
    SOME THEORY I've been doing some reading on tcp TIME-WAIT (here and there) and what I read is that it's a value set to 2 x MSL (maximum segment life) which keeps a connection in the "connection table" for a while to guarantee that, "before your allowed to create a connection with the same tuple, all the packets belonging to previous incarnations of that tuple will be dead". Since segments received (apart from SYN under specific circumstances) while a connection is either in TIME-WAIT or no longer existing would be discarded, why not close the connection right away? Q1: Is it because there is less processing involved in dealing with segments from old connections and less processing to create a new connection on the same tuple when in TIME-WAIT (i.e. are there performance benefits)? If the above explanation doesn't stand, the only reason I see the TIME-WAIT being useful would be if a client sends a SYN for a connection before it sends remaining segments for an old connection on the same tuple in which case the receiver would re-open the connection but then get bad segments and and would have to terminate it. Q2: Is this analysis correct? Q3: Are there other benefits to using TIME-WAIT? SOME PRACTICE I've been looking at the munin graphs on a production server that I administrate. Here is one: As you can see there are more connections in TIME-WAIT than ESTABLISHED, around twice as many most of the time, on some occasions four times as many. Q4: Does this have an impact on performance? Q5: If so, is it wise/recommended to reduce the TIME-WAIT value (and what to)? Q6: Is this ratio of TIME-WAIT / ESTABLISHED connections normal? Could this be related to malicious connection attempts?

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  • Scriptable BitTorrent clients?

    - by James McMahon
    In an effort further automate all the little computer house keeping tasks that can waste my time I am looking into BitTorrent clients that have the ability to script common tasks. I've done some Googling and it looks like Transmission might have some of said such capabilities, but there site wasn't very clear on the details. Things I am looking to do; Prioritize and label torrents based on trackers Set seed length based on trackers and filesize Set additional seed time when a torrent's seed time expires based on a number of factors, like time spent seeding, remaining disk space and ratio. Move torrents to appropriate places post seeding based on labels and tracker Basically, while I could Python or Bash script things like moving torrents around and other simple actions, I need away to talk to the client to figure out things like the torrent seed time, tracker, labels, filesize, etc. Is there any client out there that would allow me to all or a subset these actions? I have access to Linux, Mac and Windows and am not tied to any particular torrent client. I am a programmer so I have no problems writing scripts, but examples of torrent scripting would also be helpful.

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  • How do I make wallpaper fit both monitors in dual monitor setup?

    - by Ben
    I am deploying some custom corporate wallpaper as part of a Windows 7 rollout. Some people will be using dual monitors, and the additional monitors may be either 4:3 or widescreen. I want to use the same wallpaper on both screens (i.e. 2 copies of the same wallpaper, not stretched across both.) If I set the background to "stretch", it uses the aspect ratio of the primary monitor to stretch the wallpaper on both monitors. So, for example, if I have a dual monitor setup using a 4:3 TFT as primary and my (widescreen) laptop LCD as secondary - the image shows on the laptop LCD in 4:3, with a black stripe down either side. I've only noticed this as an issue with my "custom" wallpaper. Both the default MS wallpaper and the built in Lenovo wallpaper don't seem to have this issue. Is this by using "trickery" such as using an image larger than the largest resolution you will have and centering it? (i.e. so you crop out part of the image.) Or can this be done "properly"? I don't want to use 3rd party software to do this, but would happily do a bit of Powershell scripting if this would solve the issue. Thanks in advance, Ben

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  • How can I make XAnalogTV fill my screen?

    - by Breakthrough
    I recently installed xscreensaver, as well as the additional/extra screensavers. Many of the OpenGL ones function correctly, going fullscreen as expected. However, for some reason, the XAnalogTV screensaver leaves two "blank" spots on the edges of my screen. If I manually launch XAnalogTV, it displays a window, which it fills correctly. When I maximize the window, the same effect occurs: the window maximizes, but the two edges of the screen are literally "transparent". This effect also occurs when the screensaver is set to fullscreen. For these reasons, I believe the problem may be related to the aspect ratio of the screen. The edges of the screen are literally "ignored", with nothing being drawn there. Specifically, note the transition between the maximized and full-screen screenshots (with the un-drawn whitespace shrinking as the vertical height has been increased). For reference, I am running Xubuntu 12.04 on a Dell Vostro 1520 (Intel P8600, Nvidia 9300M) with a 1440 x 900 display (16:10). I have also set the GetViewPortIsFullOfLies preference to true. Is there any way to force XAnalogTV to fill my entire screen? Relevant screenshots (windowed, maximized, and full-screen, respectively):

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  • mplayer (mplayerhq.hu) repeats ending audio frames

    - by kamikatze
    mplayer (from mplayerhq.hu) on windows repeats the last few audio frames upon exit. When the video ends, before you can see Exiting... (End of file) in the command prompt, you will hear the last 1/2 second or so of the audio track again. This behavior is the same for multiple containers/codecs/soundcards Vista or Windows 7. Is there a workaround for this? My playback specs: MPlayer Sherpya-MT-SVN-r31027-4.2.5 (C) 2000-2010 MPlayer Team 150 audio & 343 video codecs Playing splash_final.wmv. ASF file format detected. [asfheader] Audio stream found, -aid 1 [asfheader] Video stream found, -vid 2 VIDEO: [WMV3] 1280x720 24bpp 1000.000 fps 6291.5 kbps (768.0 kbyte/s) ========================================================================== Opening video decoder: [dmo] DMO video codecs DMO dll supports VO Optimizations 0 1 DMO dll might use previous sample when requested Decoder supports the following formats: YV12 YUY2 UYVY YVYU RGB8 [..] Decoder is capable of YUV output (flags 0x1b) Movie-Aspect is undefined - no prescaling applied. VO: [directx] 1280x720 = 1280x720 Planar YV12 Selected video codec: [wmv9dmo] vfm: dmo (Windows Media Video 9 DMO) ========================================================================== ========================================================================== Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders AUDIO: 44100 Hz, 2 ch, s16le, 329.8 kbit/23.37% (ratio: 41221-176400) Selected audio codec: [ffwmav2] afm: ffmpeg (DivX audio v2 (FFmpeg)) ========================================================================== AO: [dsound] 44100Hz 2ch s16le (2 bytes per sample) Starting playback...

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  • This .mpg video clip doesn't play well

    - by Roey
    I've installed K-lite mega codec pack v6.9.0 with playback essentials without player. My default and only media player is windows media player. here are the clip's media info: General Complete name : D:\Users\Roey\Downloads\B384MV.mpg Format : MPEG-PS File size : 273 MiB Duration : 4mn 59s Overall bit rate : 7 643 Kbps Video ID : 224 (0xE0) Format : MPEG Video Format version : Version 2 Format profile : Main@High Format settings, BVOP : No Format settings, Matrix : Default Format settings, GOP : M=1, N=15 Duration : 4mn 57s Bit rate mode : Variable Bit rate : 7 363 Kbps Nominal bit rate : 9 000 Kbps Width : 1 920 pixels Height : 1 080 pixels Display aspect ratio : 16:9 Frame rate : 25.000 fps Color space : YUV Chroma subsampling : 4:2:0 Bit depth : 8 bits Scan type : Progressive Compression mode : Lossy Bits/(Pixel*Frame) : 0.142 Stream size : 261 MiB (96%) Audio ID : 192 (0xC0) Format : MPEG Audio Format version : Version 1 Format profile : Layer 3 Mode : Joint stereo Duration : 4mn 59s Bit rate mode : Constant Bit rate : 128 Kbps Channel(s) : 2 channels Sampling rate : 44.1 KHz Compression mode : Lossy Stream size : 4.56 MiB (2%) Menu When I play it there is no sound (just a little "kahhhh" noise every 10-20 seconds) and the frames are moving very slow - it "jumps" frames. A blue tray icon [FFa] "ffdshow audio decoder" pops with the following details: Input:MP3, stereo, 44100 Hz (libavocodec) Output:PCM, stereo, 44100 Hz, 16-bit integer Any help will be much appreciated. Thanks

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  • Netflix streaming stops loading at 98% on Revo 3700

    - by Martin Harris
    I'm trying to stream Netflix on an Acer Revo 3700 running Windows 7 Home Premium, but it hangs on the loading screen at 98% (after it has formatted the player to the right aspect ratio and added the controls, but before the video starts) with no error messages or failures. I have two other machines on the same network, one running Windows 7 Home Premium and another running XP, which both stream faultlessly. Things I have tried: Both a wired and wireless connection to the router Upgrading the video and audio drivers IE, Chrome and Firefox Boxee software Connecting with a VGA cable instead of HDMI (in case it is a HDCP thing) Uninstalling and reinstalling Silverlight. Getting someway into loading a HD movie and turning "Allow HD" off Does anyone know what Netflix is doing at the 98% load mark? Are there any log files? Anything else worth trying? Full disclosure: I'm using Netflix from the UK through a US based VPN. I've tried multiple VPNs and the problem is exactly the same, also the other machines on the same network through the same VPN work fine so I don't think this is the issue, but it might be a factor. The region check happens at around 7% and I get past that.

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  • Why is MySQL table_cache full but never used

    - by Jeremy Clarke
    I have been using the tuning-primer.sh script to tune my my.cnf settings. I have most things working well but the part about TABLE CACHE makes no sense: TABLE CACHE Current table_cache value = 900 tables. You have a total of 0 tables You have 900 open tables. Current table_cache hit rate is 1% , while 100% of your table cache is in use. You should probably increase your table_cache When I do SHOW STATUS; I get the following table-related numbers: Open_tables = 900 Opened_tables = 0 It seems like something is going wrong. I have some extra memory I could use on increasing the table_cache size, but my sense is that the 900 tables already available aren't doing anything, and increasing it will just waste more energy. Why might this be happening? Are there other settings that could cause all my table_cache slots to be used even though there are no hits to them? I have 150 max connections and probably no more than 4 tables per join, FWIW. Here is the tuner script output for temp tables, which I've also been tuning: TEMP TABLES Current max_heap_table_size = 90 M Current tmp_table_size = 90 M Of 11032358 temp tables, 40% were created on disk Perhaps you should increase your tmp_table_size and/or max_heap_table_size to reduce the number of disk-based temporary tables. Note! BLOB and TEXT columns are not allow in memory tables. If you are using these columns raising these values might not impact your ratio of on disk temp tables.

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  • Slowdown upon router/modem setup change

    - by Ollie Saunders
    I’ve been using a Belkin FSD7632-4 modem router to connect to my TalkTalk provided ADSL internet connection for some time and been pretty happy with it. Recently, however, the connection has been failing and I decided to get a ASUS RT-N16 instead, which is also a much more capable router generally. The ASUS RT-N16 doesn’t come with a modem built-in so I purchased as Zoom modem as well. I’ve set them both up and am using them to post this message. But I’m a bit miffed to find that I get a significantly and consistently slower downstream rate from the new configuration than with the old Belkin. Belkin modem router: downstream: 3.45 mbps upstream: 0.73 mbps ASUS router + Zoom modem: downstream: 2.71 mbps upstream: 0.66 mbps Any ideas why this is? The really weird thing about this is that the Zoom supports ADSL2 and ADSL2+ but I don’t think the old Belkin does. At first I thought it might be due to the Zoom modem being limited to PPPoE instead of PPPoA, which my ISP supports, but then I tried using PPPoE with the Belkin and that still gave a high speed. I’m using VC-Mux encapsulation with both. VPI of 0 and VCI of 38. I pulled this data off the Zoom: Mode: ADSL2 Line Coding: Trellis On Status: No Defect Link Power State: L0 Downstream Upstream SNR Margin (dB): 12.3 11.8 Attenuation (dB): 43.0 24.9 Output Power (dBm): 12.9 0.0 Attainable Rate (Kbps): 3936 844 Rate (Kbps): 3194 840 MSGc (number of bytes in overhead channel message): 59 10 B (number of bytes in Mux Data Frame): 99 14 M (number of Mux Data Frames in FEC Data Frame): 2 16 T (Mux Data Frames over sync bytes): 1 8 R (number of check bytes in FEC Data Frame): 8 8 S (ratio of FEC over PMD Data Frame length): 1.9833 9.0594 L (number of bits in PMD Data Frame): 839 219 D (interleaver depth): 32 2 Delay (msec): 15 4 Super Frames: 15808 14078 Super Frame Errors: 0 4294967232 RS Words: 513778 111753 RS Correctable Errors: 126 4294967238 RS Uncorrectable Errors: 0 N/A HEC Errors: 0 4294967279 OCD Errors: 0 0 LCD Errors: 0 0 Total Cells: 1920175 237597 Data Cells: 205993 392 Bit Errors: 0 0 Total ES: 0 0 Total SES: 0 0 Total UAS: 34 0

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  • ffmpeg: video file played OK on Ubuntu, but no sound on XP

    - by Andy Le
    I created a video clip using ffmpeg (vcodec: mpeg2video, acodec: AC3 5.1). The file can be played normally on Ubuntu, but when I play it on an XP machine, there is no sound. I can play AC3 files and other movies with AC3 sound. I already tried many codec packs and many players. When I compare the MediaInfo tab of the Properties window of the file with another playable movie, I see that the Audio Identifier of the audio stream in my file is 0x80 while it is 0x02 in the other movie. So I guess that's why players on XP can't recognize the audio codec. When I use an MKV container instead of MPEG (still mpeg2video codec), then the result is OK on both Ubuntu and XP (with the correct Audio ID). I really need MPEG though. Any idea? This is the command I used: ~/ffmpeg/ffmpeg/ffmpeg -loop_input \ -t 97 -r 30000/1001 -i v%4d.tga -i final.ac3 \ -vcodec mpeg2video -qscale 1 -s 400x400 -r 30000/1001 \ -acodec copy -y out6.mpeg 2 This is the output of mediainfo (on Ubuntu): General Complete name : out6.mpeg Format : MPEG-PS File size : 6.86 MiB Duration : 1mn 37s Overall bit rate : 593 Kbps Video ID : 224 (0xE0) Format : MPEG Video Format version : Version 2 Format profile : Main@Main Format settings, BVOP : No Format settings, Matrix : Default Format_Settings_GOP : M=1, N=12 Duration : 1mn 37s Bit rate mode : Variable Bit rate : 122 Kbps Width : 400 pixels Height : 400 pixels Display aspect ratio : 1.000 Frame rate : 29.970 fps Resolution : 8 bits Colorimetry : 4:2:0 Scan type : Progressive Bits/(Pixel*Frame) : 0.025 Stream size : 1.41 MiB (21%) Audio ID : 128 (0x80) Format : AC-3 Format/Info : Audio Coding 3 Duration : 1mn 36s Bit rate mode : Constant Bit rate : 448 Kbps Channel(s) : 6 channels Channel positions : Front: L C R, Side: L R, LFE Sampling rate : 44.1 KHz Stream size : 5.18 MiB (75%)

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