Search Results

Search found 10018 results on 401 pages for 'sample rate'.

facebook twitter digg google delicious technorati stumbleupon myspace wordpress linkedin gmail igoogle windows live tumblr viadeo yahoo buzz yahoo mail yahoo bookmarks favorites email print

Page 52/401 | < Previous Page | 48 49 50 51 52 53 54 55 56 57 58 59  | Next Page >

  • Simulating a low-bandwidth, high-latency network connection on Linux

    - by Justin L.
    I'd like to simulate a high-latency, low-bandwidth network connection on my Linux machine. Limiting bandwidth has been discussed before, e.g. here, but I can't find any posts which address limiting both bandwidth and latency. I can get either high latency or low bandwidth using tc. But I haven't been able to combine these into a single connection. In particular, the example rate control script here doesn't work for me: # tc qdisc add dev lo root handle 1:0 netem delay 100ms # tc qdisc add dev lo parent 1:1 handle 10: tbf rate 256kbit buffer 1600 limit 3000 RTNETLINK answers: Operation not supported How can I create a low-bandwidth, high-latency connection, using tc or any other readily-available tool?

    Read the article

  • ffmpeg add two audio streams to video

    - by Tossin Hausen
    I tried this: ffmpeg -i /sdcard/video/transcode/video.avi -map 0:0,0 -i /sdcard/video/transcode/first.mp3 -map 1:0,1 -i /sdcard/video/transcode/second.mp3 -map 2:0,2 -acodec copy -vcodec py /sdcard/video/transcode/Output.avi to add two audio streams to one video file. But ffmpeg says the number of mappings should match the number of output streams. What is wrong here? I'm trying to work with an Android build of FFmepg "ffmpeg for android beta". "Does not work" means that this uncommunicative Android build of FFmpeg just stops without giving any error message. The -codec copy option does not work with this build. Now I tried the same set of files with the FFmpeg called command line tool that comes with Ubuntu 10. Something (can't say where it is from). The -codec copy option does not work with this FFmpeg too. Here the complete output: m30x:~/movie/Film$ ffmpeg -i input.avi -i first.mp3 -i second.mp3 -map 0 -map 1 -map 2 -acodec copy -vcodec copy output.avi FFmpeg version SVN-r0.5.9-4:0.5.9-0ubuntu0.10.04.1, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.9-0ubuntu0.10.04.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Jun 12 2012 16:27:34, gcc: 4.4.3 [NULL @ 0x93cfd10]looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (30000/1) -> 25.00 (25/1) Input #0, avi, from 'input.avi': Duration: 01:30:33.00, start: 0.000000, bitrate: 901 kb/s Stream #0.0: Video: mpeg4, yuv420p, 576x432, 25 tbr, 25 tbn, 30k tbc Input #1, mp3, from 'first.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #1.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Input #2, mp3, from 'second.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #2.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Number of stream maps must match number of output streams Merging only one audio stream with the video stream works with Ubuntu and Android version of FFmpeg. Here the complete output: ffmpeg -i input.avi -i first.mp3 -map 0 -map 1 -acodec copy -vcodec copy output.avi FFmpeg version SVN-r0.5.9-4:0.5.9-0ubuntu0.10.04.1, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.9-0ubuntu0.10.04.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Jun 12 2012 16:27:34, gcc: 4.4.3 [NULL @ 0x9bfad10]looks like this file was encoded with (divx4/(old)xvid/opendivx) -> forcing low_delay flag Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (30000/1) -> 25.00 (25/1) Input #0, avi, from 'input.avi': Duration: 01:30:33.00, start: 0.000000, bitrate: 901 kb/s Stream #0.0: Video: mpeg4, yuv420p, 576x432, 25 tbr, 25 tbn, 30k tbc Input #1, mp3, from 'first.mp3': Duration: 01:30:32.84, start: 0.000000, bitrate: 63 kb/s Stream #1.0: Audio: mp3, 22050 Hz, stereo, s16, 64 kb/s Output #0, avi, to 'output.avi': Stream #0.0: Video: mpeg4, yuv420p, 576x432, q=2-31, 90k tbn, 25 tbc Stream #0.1: Audio: libmp3lame, 22050 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #1.0 -> #0.1 Press [q] to stop encoding frame= 6157 fps=6156 q=-1.0 size= 31667kB time=246.28 bitrate=1053.3kbits/s Do you have an idea why it does not work with two audio streams? By the way, ffmpeg -i input_with_first_audio_stream.avi -i second.mp3 -acodec copy -vcodec copy output_two_audio_streams.avi -newaudio works with both versions of ffmpeg that I use, but the first audio stream is played too fast (x10 or more), while the second audio stream is played correct. Many thanks in advance and sorry for my unconventional question and outdated versions of ffmpeg. But I am a lamer and it is not so easy for me to compile from the source (especially for the Android version). I will try to compile an up to date version of ffmpeg with Ubuntu, but I don't have much free time.

    Read the article

  • php-fpm or nginx: bad gateway

    - by John Tate
    I'm getting a bad gateway error all the sudden for a site. I didn't change the configuration for the site, I just added a new server config where I put them under /etc/nginx/servers and it stopped working. The new server works, and there is no conflict between the php-fpm listen addresses. server { listen 80; server_name obfuscated.onion; location = / { root /var/www/sites/obfuse; index index.php; } location / { root /var/www/sites/obfuse; index index.php; if (!-f $request_filename) { rewrite ^(.*)$ /index.php?q=$1 last; break; } if (!-d $request_filename) { rewrite ^(.*)$ /index.php?q=$1 last; break; } } error_page 404 /index.php; location ~* ^.+.(jpg|jpeg|gif|css|png|js|ico)$ { root /var/www/sites/obfuse; access_log off; expires 30d; } location ~ \.php$ { fastcgi_pass 127.0.0.1:9000; fastcgi_index index.php; fastcgi_param SCRIPT_FILENAME /var/www/sites/obfuse$fastcgi_script_name; fastcgi_param QUERY_STRING $query_string; fastcgi_param REQUEST_METHOD $request_method; fastcgi_param CONTENT_TYPE $content_type; fastcgi_param CONTENT_LENGTH $content_length; fastcgi_param PATH_INFO $fastcgi_script_name; include fastcgi_params; } } There is nothing unusual in php-fpm's log even when I raised the level to debug. [24-Jun-2013 09:10:37.357943] DEBUG: pid 6756, fpm_scoreboard_init_main(), line 40: got clock tick '100' [24-Jun-2013 09:10:37.358950] DEBUG: pid 6756, fpm_event_init_main(), line 333: event module is kqueue and 1 fds have been reserved [24-Jun-2013 09:10:37.358978] NOTICE: pid 6756, fpm_init(), line 83: fpm is running, pid 6756 [24-Jun-2013 09:10:37.359009] DEBUG: pid 6756, main(), line 1832: Sending "1" (OK) to parent via fd=5 [24-Jun-2013 09:10:37.389215] DEBUG: pid 6756, fpm_children_make(), line 421: [pool cyruserv] child 22288 started [24-Jun-2013 09:10:37.391343] DEBUG: pid 6756, fpm_children_make(), line 421: [pool cyruserv] child 21911 started [24-Jun-2013 09:10:37.391914] DEBUG: pid 6756, fpm_event_loop(), line 362: 5776 bytes have been reserved in SHM [24-Jun-2013 09:10:37.391941] NOTICE: pid 6756, fpm_event_loop(), line 363: ready to handle connections [24-Jun-2013 09:10:38.393048] DEBUG: pid 6756, fpm_pctl_perform_idle_server_maintenance(), line 379: [pool cyruserv] currently 0 active children, 2 spare children, 2 running children. Spawning rate 1 [24-Jun-2013 09:10:39.403032] DEBUG: pid 6756, fpm_pctl_perform_idle_server_maintenance(), line 379: [pool cyruserv] currently 0 active children, 2 spare children, 2 running children. Spawning rate 1 [24-Jun-2013 09:10:40.413070] DEBUG: pid 6756, fpm_pctl_perform_idle_server_maintenance(), line 379: [pool cyruserv] currently 0 active children, 2 spare children, 2 running children. Spawning rate 1 I don't know why this has started happening, but the logs are not telling me anything. Please ask for more information than this, you'll probably need it.

    Read the article

  • Why does CPU processing time matter when compared to real wall clock time?

    - by PeanutsMonkey
    I am running the command time 7zr a -mx=9 sample.7z sample.log to gauge how long it takes to compress a file larger than 1GB. The results I get are as follows. real 10m40.156s user 17m38.862s sys 0m5.944s I have a basic understanding of the difference but don't understand how this plays a role in the time in takes to compress the file. For example should I be looking at real or user + sys?

    Read the article

  • how to record mic input and pipe the output to another program

    - by acrs
    Hi everyone Im trying to follow a tutorial on generating truly random bits How To Generate Truly Random Bits This is the command from the tutorial but it does not work rec -c 1 -d /dev/dsp -r 8000 -t wav -s w - | ./noise-filter >bits I know i can record my mic input using rec -c 1 no.wav this is the command i tried using rec -c 1 -r 8000 -t wav -s noise.wav | ./noise-filter >bits but i get root@xxc:~/cc# rec -c 1 -r 8000 -t wav -s noise.wav - | ./noise-filter >bits rec WARN formats: can't set sample rate 8000; using 48000 rec FAIL sox: Input files must have the same sample-rate I have complied noise-filter noise-filter I think the tutorial is using an older version of SOX and REC I'm using sox: SoX v14.3.2 on Ubuntu 12.04 server Can someone please help me ?

    Read the article

  • 1080 HD video display - Windows 7

    - by blasteralfred
    I am running Windows 7 in my Dell Studio 1555 Laptop. I have ATI Mobility Radeon HD 4570 graphics card inside. When I try to watch a video, I can see only some disturbed texture. I can hear the audio and no prob with that. The I can see the video frames peeking sometimes but is pretty slower. I have DirectX (directx_feb2010_redist) and Win7 codecs installed. Here are some screenshots; The video details are; Name: video.mp4 Frame width: 1920 Frame height: 1080 Frame rate: 29 fps data Rate: 7958 kbps Total bitrate: 8052 kbps Just comment if you want more info.

    Read the article

  • how to limit upload bandwidth per user in linux?

    - by Gihan Lasita
    Can anyone provide the tc command to limit upload bandwidth per user in Debian Lenny? I found that to mark packets per user with iptables I can use the following command iptables -t mangle -A OUTPUT -p tcp -m owner --uid-owner testuser -j MARK --set-mark 500 but I have no idea how to use tc update by running following commands, i managed to limit testuser upload bandwidth to 10Mbit iptables -t mangle -N HTB_OUT iptables -t mangle -I POSTROUTING -j HTB_OUT iptables -t mangle -A HTB_OUT -j MARK --set-mark 30 iptables -t mangle -A HTB_OUT -m owner --uid-owner testuser -j MARK --set-mark 10 tc qdisc replace dev eth0 root handle 1: htb default 30 tc class replace dev eth0 parent 1: classid 1:1 htb rate 10Mbit burst 5k tc class replace dev eth0 parent 1:1 classid 1:10 htb rate 10Mbit ceil 10Mbit tc qdisc replace dev eth0 parent 1:10 handle 10: sfq perturb 10 tc filter add dev eth0 parent 1:0 prio 0 protocol ip handle 10 fw flowid 1:10 now the problem is, i do not want to limit testuser's FTP bandwidth but by running above commands FTP speed also limited to 10Mbit. Regards

    Read the article

  • SoX on Windows 7 64-Bit Outfile Missing

    - by Christian
    I have come across the strangest problem when trying to run sox.exe on my Windows 7 installation. Whenever I try and record audio it works without any issues but it will not output an audio file. The crazy thing is that when I use the play command it successfully plays what I just recorded. Has anyone ever heard of this happening? Here are the commands (and output) that I'm using: C:\Program Files (x86)\Vox\sox-14-4-0>sox -d test.wav trim 0 00:05 Input File : 'default' (waveaudio) Channels : 2 Sample Rate : 48000 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM In:0.00% 00:00:05.03 [00:00:00.00] Out:240k [ | ] Clip:0 Done. C:\Program Files (x86)\Vox\sox-14-4-0>play test.wav test.wav: File Size: 960k Bit Rate: 1.54M Encoding: Signed PCM Channels: 2 @ 16-bit Samplerate: 48000Hz Replaygain: off Duration: 00:00:05.00 In:100% 00:00:05.00 [00:00:00.00] Out:240k [ | ] Clip:0 Done. Am I losing my mind or is something up here?

    Read the article

  • On Windows 7, how can I tell if a recording is multi-channel without third party tools? [migrated]

    - by engineerchuan
    A customer has an audio that is confidential and can't send it to me. He also would not like to install other tools. He has a basic Windows 7 install. Is there any way to tell whether the recording is one channel or two channel? Normally, I would just get the audio and soxi it. Or, I would tell him to install Audacity or equivalent sound editor and open it up. I also thought that if you right clicked and looked at the size, bit rate, and length, you could get number of channels but bit rate already factors in number of channels. Sorry I'm not giving you a lot to work with.

    Read the article

  • Throttling apache downloads selectively

    - by Synchro
    I have a linux box running Debian Sarge (old I know) and apache 2.0.54. It serves two kinds of files - regular web pages and small images, and a lot of large podcast mp3s. The podcast downloads swamp the connection and make the rest of the site unresponsive, so I'm looking to throttle the data transfer rate (not the request rate) of just the podcasts. I've set up haproxy using this technique which does what it says it will, but solves a different problem - even only 5 simultaneous podcast downloads is enough to saturate the link. In a perfect world, haproxy would support per-connection throttling, but it doesn't. So far I've looked at mod_bw (won't compile for me, seems unsupported), mod_cband (unsupported, widely reported as problematic) and iptables using tc. The iptables approach would allow me to throttle things, but would not be at all selective, slowing down everything on the server, not just the podcasts, so would just move the bottleneck without changing overall behaviour. Ideas?

    Read the article

  • Are there any command line utilities which can calculate and/or limit how fast a pipe is running?

    - by stsquad
    I'm doing some basic stress testing of a Linux kernel network IWF with netcat. The set-up is fairly simple. On the target side: nc -l -p 10000 > /dev/null And on my desktop I was running: cat /dev/urandom | nc 192.168.0.20 10000 I'm using urandom for some poor-mans fuzz testing. However I find that even at this rate I can break something quite quickly. EDIT So I've been playing with trickle to rate limit how fast I'm generating data: cat /dev/urandom | trickle -u 10 nc 192.168.0.20 10000 But it's hard to tell if this is working. What would be really useful is a the pv equivilent of trickle that can work with pipes.

    Read the article

  • How to force 640*480@60Hz screen resolution on xubuntu 12.04

    - by c2h2
    It seems xubuntu won't be able to correctly set resolution at 640*480@60Hz at its Display settings. And I am unable to correctly my super small 6.4 inch Mitsubishi VGA panel via VGA cable. I have tried to hack both X11 conf /etc/X11/Xorg.conf and xfce4 conf, but all the document I can find is outdated. and conf files are changed into other location. Can someone give me a hand and I'll mark correct for other people to use? Thanks! EDIT: The board is an Intel Atom D2700, gpu is SGX545. I tried to use xrandr --output default --mode 640*480 It seems works fine, but refresh rate is 75Hz, but the screen only suports 60Hz So I used xrandr --output default --mode 640*480 --rate 60 but it give error: xrandr: Failed to get size of gamma for output default Can anyone pointing any directions?

    Read the article

  • Data transfer speed to USB storage connected to wifi router very slow

    - by RonakG
    Here is my setup. A Linksys Cisco E3200 wifi router. A MacbookPro running OS X Lion 10.7.4. A Seagate GoFlex 1TB hard drive connected to wifi router via the USB port. When I try to transfer data from my MBP to the HDD, the data transfer rate is very low. I'm getting around 3MB/s write speed. This is very slow compared to the speed I get when HDD is directly connected to the MBP. The HDD is NTFS formatted. And the router provides access to HDD using Samba share. So I connect to the HDD using smb://. What is the limiting factor here affecting the data transfer rate?

    Read the article

  • On Windows 7, how can I tell if a recording is multi-channel without third party tools?

    - by engineerchuan
    A customer has an audio that is confidential and can't send it to me. He also would not like to install other tools. He has a basic Windows 7 install. Is there any way to tell whether the recording is one channel or two channel? Normally, I would just get the audio and soxi it. Or, I would tell him to install Audacity or equivalent sound editor and open it up. I also thought that if you right clicked and looked at the size, bit rate, and length, you could get number of channels but bit rate already factors in number of channels. Sorry I'm not giving you a lot to work with.

    Read the article

  • Why is the pavucontrol level indicator jumping while nothing plays?

    - by EnterTheLiquidToasterFamily
    The level indicator in the screenshot does jump around even if nothing is playing. The indicator also reasonably represents sound levels when music is playing. I dont have any mediaservers running or noisy browsertabs open. Also no mic connected. When I turn the volume to max in software and on the amp, there is no noise from the speakers at all. Played music is loud and not distorted. Hardware: Realtek ALC889 over optical audio connector to a generic amp. Software: Debian Wheezy with latest backport kernel 3.14 (same thing on wheezy 3.2 stock), wheezy pulseaudio, xfce session, a custom asound.conf that enables pulseaudio to push sound over optical port. /etc/asound.conf pcm.a52 { @args [CARD] @args.CARD { type string } type rate slave { pcm { type a52 bitrate 448 channels 6 card $CARD } rate 48000 #required somehow, otherwise nothing happens in PulseAudio } }

    Read the article

  • Do I use the FV function in Excel correctly?

    - by John
    My task: Create a table: Calculate what the revenues of e-trading will be after five years at 15 percent interest rate if we now have 15 000 EUR. Use the FV function from the Financial Group in Excel. My resolution: =FV( 15%; 5; 0; -15000). My question: Is it correct? I know the task lacks information whether the interest rate is per month or per year. I calculate it as 'per year'. My question is orientated more on the usage of the FV function. I, for example, do not understand why '-15000' and not '15000'. Also why the third parameter has to be 0? Maybe I do it wrong. Please help me solve it! Thanks in advance.

    Read the article

  • Glassfish Virtual Servers with Different Contents

    - by Nikael Vergara
    I have been searching google for hours trying to find out whether different virtual servers in glassfish 3.1 could contain different contents. For example there are two virtual servers named VS1 and VS2. I added a sample application on VS1. Is the sample application I added on VS1 can be edited or seen on the admin console of VS2? In this feature possible in glassfish? Any help will be appreciated.

    Read the article

  • Is Protune for video only or may be used for photo too?

    - by Green
    I have Hero3+ Black Edition. I can't understand if Protune is for video only or may be applied for photos? The manual says it is both for video and for photo (page 35): High-Quality Image Capture Protune’s high data rate captures images with less compression, giving content creators higher quality for professional productions. Film/TV Rate Standard While shooting in Protune, you have the option of recording video in cinema quality 24 fps to easily intercut GoPro content with other source media without the need to perform fps conversion. But at the same time their site says that Protune is for video only: To record Protune footage, you’ll need to turn Protune ON in your camera’s settings menu. What for is Protune? Photo? Video? Or both?

    Read the article

  • Network speeds being report as 4x higher than actual in Windows 7 SP1

    - by Synetech
    Ever since installing Windows 7 SP1, I have noticed that all programs that display my network transfer rate have been exactly 4x higher than they actually are. For example, when I download something from a high-bandwidth web site or through torrents with lots of sources, the download rate indicated is is ~5MBps (~40Mbps) even though my Internet connection has a maximum of only 1.5MBps (12Mbps). It is the same situation with the upstream bandwidth: the connection maximum is 64KBps, but I’m seeing up to 256KBps. I have tried several different programs for monitoring bandwidth throughput and they all give the same results. I also tried different times and different days, and they always show the rate as being four times too high. My initial thought was that my ISP had increased the speeds (without my noticing), which they have done before. However, I checked my ISP’s site and they have not increased the speeds. Moreover, when I look at the speeds in the program actually doing the transfer (eg Chrome, µTorrent, etc.), the numbers are in line with the expected values at the same time that bandwidth monitoring programs are showing the high numbers. The only significant change (and pretty much the only change at all) that has occurred to my system since the change was the installation of SP1 for Windows 7. As such, it is my belief that some sort of change exists in SP1 whereby software that accesses the bandwidth via a specific API receives (erroneously?) high numbers while others that have access to the raw data continue to receive the correct values. I booted into Windows XP and downloaded some things via HTTP and torrent and in both cases, the numbers were as expected (like they were in Windows 7 before installing SP1). I then booted back into 7SP1 and once again, the numbers were four times higher than possible. Therefore it is definitely something in SP1 that has changed how local bandwidth is calculated/returned. There is definitely something wonky with Windows 7 SP1’s network speed calculation. I tried Googling this, but (for multiple reasons), have had a difficult time finding anything relevant. Has anybody else noticed this behavior? Does anybody know of any bugs or changes in SP1 that could account for it?

    Read the article

  • Data transfer is extrem slow after partitioning extern usb drive

    - by user125912
    I bought an extern usb 3.0 drive with 500 gb capacity. OS is Windows 7. I use it with an usb 2.0 slot, no prob. Initially I used it without making several partitions and it was fast as hell. Then I had the great idea to make partitions, one for programs, one for data and one for backup. I chose the free EASEUS Partition Master 9.1.1. and ended up with these partitions: F:Apps, primary, NTFS, 100gb H:Data, logic, NTFS, 250gb B:Backup, logic, NTFS, 150gb THE PROBLEM: When I copy files from C: to F: I get a transfer rate of about 100 KB/S ! When I copy files from C: to H: I get a transfer rate of about 4 MB/S ! thats all muuuch to slow, slower then before. What can I do to speed the shit up? Thanks in advance!

    Read the article

  • Network Performance issue

    - by qubemarker
    We have three Ubuntu 10.04 servers. One server is a storage server and the other two servers are configured as clients. The storage server has a good amount of capacity and it is integrated with windows Active directory server for Authentication. I am uploading some video files from both clients to the server and when I am uploading data from any one client alone I get about 26 MB/s data transfer rate. When I upload data from both the clients simultaneously I am only getting about 8 MB/s from each client. I have gigabit ethernet cards in all of the servers and a L2 Managed gigabit switch for connectivity. I don’t know why the data transfer rate is decreasing so much in simultaneous read and write. I have tried all of the TCP stack related settings suggested here. Can any assist with getting better read/write performance out of this setup? Any help is appreciated.

    Read the article

facebook twitter digg google delicious technorati stumbleupon myspace wordpress linkedin gmail igoogle windows live tumblr viadeo yahoo buzz yahoo mail yahoo bookmarks favorites email print

< Previous Page | 48 49 50 51 52 53 54 55 56 57 58 59  | Next Page >