I am trying to find out how I can estimate the IP Layer bitrate I would get from an ethernet link? Say at 100 Mbits/s or 200 Mbits/s (powerline)? Is there a formula that I could use to calculate that?
im trying to send pages of 4096 bytes from kernel layer of server to kernel layer of client over a network. previously i tried the foll. code , for data less than a 100 bytes it worked fine , but for something larger than that the computer hangs......(even the dmesg's wont say why) i also wanted to know how we could use the 'sendpage' function to solve this problem.
I have 1 router 2 pcs. We had to move one pc away from the router so I bought a wifi card. Everything went fine, the card works and I have internet but the network is incredible slow.
I want to write a new protocol on transport layer. These are the features that new protocol should include:
-Real time -Concurrent transfer -Multipath and Multihoming
the reason why i should write this protocol is in order to transport encoded video stream over 3G link. When i investigated TCP UDP or SCTP source codes, i couldnot understand them at all. What ı want is to help me about steps of the protocol writing or guideline about it. How should i start, is there any book or source about it (i couldnot find over my internet research).
I'm looking for a way (kernel patches, configuration, etc) to bond multiple network interfaces together but for limited purposes. Here's the setup. Machines A, B, C, and D each have 4 NICs, each of which are on separate unmanaged switches. The connections are made in a corresponding way. e.g. eth0 of each machine are connected via switch 0, eth1 are connected via switch 1, etc. There are also other machines which have only one NIC and are connected to switch 0 only. All NICs for A/B/C/D and the switches are gigabit speed. The remaining machines have a low traffic level. Machines A/B/C/D need the extended bandwidth. And this bandwidth need usually involves only one connection at a time.
E.g. machine A transferring files to machine C with no other traffic going on. The speed need is to cut the transfer times from several hours to few hours (such as 8 hours to 2 hours). Transfers of up to a few terabytes at a time are involved. IEEE 802.1AX won't accomplish this. It requires special support from a single switch that all connection go to (raising costs and reducing reliability). Also, from technical details of 802.1AX, it appears that a decision process is made for which traffic goes over which physical link based on destination information. It's unclear what impact this will have, but it looks like at least a single TCP connection cannot use all physical links.
And possibly all traffic from host A to host B is limited to a physical link (not any better than a round robin of crossover cables). What I am looking for is something that works entirely on an end-to-end basis within a LAN. If it works at the link layer, that could be OK as long as it doesn't have the limitations of 802.1AX. Working at the IP layer would be OK, too (as I can already envision the logic of how to make that work). This might be an experimental patch to the Linux kernel if anyone has tried it. I have not dug into kernel source to see what might be in there, yet, but will eventually do that if there isn't a patch already available.
Our ISP provides us with Layer 2 hardware (Modem?/Switch? (Hatteras)) for our leased line and internet access. I have been told that I should set up a VLAN capable router to separate the internet traffic from the internal traffic. I found that linux is capable of VLAN routing. Nice!
I have setup opensuse 10.3 put in two NICs and did the following vconfig add eth0 10 vconfig add eth0 20 ifconfig public.ip.add.ress netmask 255.255.255.252 eth0.10 up ifconfig 192.168.0.1 netmask 255.255.255.0 eth0.20 up
Plugged this NIC into the HATTERAS hardware (with a straight cable), and thought that this way I should be able to ping the public gateway or any ip out on the internet. My ISP is telling me that I should create a VLAN trunk to be able to 'use the internet', but as I understand in linux if I create any number of VLANs on one NIC they are already trunked. I also got the info, that the traffic is tagged, and I can separate the traffic reading the tags. I already read that some NICs are not able to VLAN because they are not able to handle the increased packet size.
Also that the MTU setting is important (dono' the exact value though, only that its important). I thought that a linux machine can act as a router and firewall in such a case, because proprietary switches/routers use linux as embedded os. This is my first meeting with VLAN so if this whole post does not make any sense or you think that I just need to RTFM more then tell me! I also have some (3) Dlink 3226 Layer2 switches around, but I think it would be waste to use a 24 port switch on this subject.
I was wondering how to install the Link Layer Topology on CentOS 5. I have installed the lld2d daemon from a Debian how to after compiling it. It seems to be running but it is not talking to the Windows 7 network mapping.
I have a CentOS5 server with a 1tb hard drive.There is only 80gb of data on that huge drive and now I want to make a bare metal recovery backup using AcronisMy question is, how can I estimate the amount of time the backup will take and the size of the image file? Is it based on the size of my drive or is it based on the amount of data on the drive?
We run redundant switches that two nic's on each server connect to. We also run bonding on our servers. Because we have two switches, we can't run lacp or anything. If a switch goes into a crashed state where it doesn't pass traffic but still provides link, bonding thinks the interface is still up and thus will still send traffic through it. Does anybody know a better way to configure the fail over of the interface? This would be a similar situation to somebody using a media converter.
In fact the bitrate should be 139 or possibly 132. So assume that is what is wrong. Have assiduously coated myself with nourishing smegma and the like but am unable to work out how to set the bitrate in this case.
I have the same mp3 on two different distros, Fedora 14 and Ubuntu 10.10, both using gnome. The md5sum of the mp3 is identical on both distros, but on Ubuntu the bitrate is 96kbps and on Fedora it is apparently 257kbps. Discovered this using RhythmBox but also appears in Nautilus browser, under the properties tag.
Kernel 2.6.21.5, Slakware 12.0 I have plenty of files like this:
Code:
$ file 23-1.mp3 23-1.mp3: Audio file with ID3 version 23.0 tag, MP3 encoding $
All I want to know about a file like this is the bitrate is has been created with. Can this data be inside the tag? What's a cli program that lets me know that information?
I am wanting to reduce around 9.9gigs of music to 8gigs to fit on my phone. I have done some snooping and noticed a program called Lame. I also noticed a code:
for x in [ `ls -1 *.mp3` ]; do lame --preset 32 $x new32-${x}; done
I was wondering if someone would be able to explain the code and how I go about converting the bitrate to 96?
The subject of this post is actually a question e.g. is there a mp3 ripper that allows the bitrate to be modified. I have looked at cdparanioa, which I believe is the foundation of most of the Linux rippers, but it does not allow modification of the bitrate. I realise I could ripper in flacc format but I have an mp3 player as does my grandaughter so I need the mp3 format. Incidentally is there a player(portable) that will play flacc encoded files?
This may be a minor irritation, but I'm puzzled by it nonetheless. I'm ripping my CDs to MP3 using Sound Juicer with the lamemp3enc plugin and following gstreamer pipeline.
All was great a couple days ago, but suddenly the files' audio properties show 48 kbps bitrate when it should be well over 200. I even changed the pipeline to use "target=1 bitrate=256" with the same result. The files are being encoded as expected and that is reflected in the file size and the Statistics view in VLC.
I have two computers running Ubuntu 10.10. One has all the latest updates, but the other has not been updated in several days. This problem is happening on the former, but not the latter. gst-inspect lamemp3enc shows both computers have the same version of the encoder.
I thought I'd put it out to the forum here before submitting a bug in launchpad.
I really need to burn my 64 kbps uncompressed A-Law PCM file to Audio CD uncompressed as is and make this 8 hour file appear on CD Players. What app does it for me? It must fit on CD! Goldwave makes a-law / u-law wav files BTW. I can do this with linux or windows with any application. tell me how or where to download. I have the wav files already.
I have a large collection of music albums sorted in folders which are named like this: "Zombie Ritual - 2004 - Night Of The Zombie Party" (%{artist} - %{year} - %{album}). I want to rename them so as to be indicative of the bitrate, for example, "Zombie Ritual - 2004 - Night Of The Zombie Party" => "Zombie Ritual - 2004 - Night Of The Zombie Party (@320)". It will be hard to do this manually. I tried to use EasyTag and Kid3 to do this, but they cannot add bitrate to tags.
Mencoder settings... I'm doing a simple transcoding of an avi file from FFmpeg MPEG4 codec to XVID MPEG-4 codec. The problem is that the audio bitrate is changing from 192 kbps to 32 kbps. The audio codec for both files is mp3 (MPEG-1 layer 3). (I'm transcoding the FFmpeg MPEG4 encoded avi because my blu-ray player doesn't see, let alone play it. Any XVID MPEG-4 encoded avi files the blu-ray player sees and plays fine). I ran mencoder trying the four settings below but the audio bitrate still comes out at 32 kbps:
When I convert files with mencoder I get the output with incorrect audio bitrate. Seems that mencoder ignores the bitrate I pass to it. Here's my script:
Result file should have 128kbps audio bitrate, but here are the results: Input file: 02_seminar.avi, 42Mb video: 432x320 00:05:10 25fps DivX5 1Mbps audio: 48KHz 00:05:10 Stereo 138Kbps mp3
howto to transcode multiple mp3 files to another bitrate? I have a lot of 320bits mp3 which should be converted to 192bits before I can play them on my car stereo.
I've been testing with the SoundConverter software and I want to write a script for it.Most music files on my pc are *.flac. But I want to convert some albums to my mp3 player with a script. Everything works fine. I do this:Code:soundconverter -b -m audio/mpeg -s .mp3 *.flacBut the quality is 128kbps.Is there any way to change the bitrate (in the terminal ofcourse)?And if this is not possible, is there an alternative that copies the tags correct like SoundConverter?
I have finally found a native Linux music player that's as light and as simple as Foobar(unfortunately Foobar under Wine is giving me problems so I needed to find a native player). creating custom columns for the player. And since I can't find any DeadBeef forums, I'm hoping people here are familiar with this player.
So, the two main custom columns I'm looking to create are Codec and Bitrate. Codec to show if the tracker is mp3, ogg, flac, etc. And Bitrate to show the bitrate of each track.
Now there's an option in DeadBeef to create columns, but I'm having trouble figuring out how to get the column to show the info that I want. I was hoping that the way you get custom columns to show up in Foobar would work in DeadBeef. But inputting %bitrate% and %codec% into the columns in DeadBeef does not work.
how to get the two columns that I'm wanting to display the info that I want?
Or at least that's what I figured, when watching videos from ....., etc., the sound in the left channel turns into distorted high frequency noise, anyone had the same issue? Is there a fix for this? Sound in other applications works just fine.EDIT: Oh, and I'm using the proprietary flash plugin, has worked just fine on my Ubuntu Studio on the exact same computer.
I am looking for an application to detect mp3 duplicates using the mp3 spectrograph (sound wave matching) or something similar because simply I have many mp3 files that are the same but with different bitrate,ID3 tags,filesize
I am looking for a layer 7 firewall. I have to redirect rtmp requests for different hostnames coming at a gateway to internal servers at LAN at their respective hostnames. Code: IPTABLES some stuff -p rtmp -hostname to server 1 like that. Or if not IPTABLES then some other feasible solution.