Actually I was starting to use i3 window manager then I just want to bring the horizontal window to vertical one as shown below. I have search and then I got the info from i3 FAQ.
-------------||||| A | B | C |||||-------------
------------|| B || A |---||| C |------------
The exact keys you would use depend on your configuration. I’ll use this config file: http://i3wm.org/docs/user-contributed… for reference and list the corresponding i3 commands. The modifier key $mod is most likely Alt or the Windows key on your computer.
If the window B is not focused, focus it via either $mod+Left or $mod+Right (command focus left or focus right).
Split the window in the vertical direction: $mod+v (command split v). There may be little or no visible difference (perhaps the bottom edge becomes visibly active).
Move focus right, to the window C, using $mod+Right (command focus right).
Move the window C to the left, using $mod+Shift+Left (move left). It should now be in the same column as the window B.
As I was using Debian based flavours and redhat based Flavours I did not find any difficult in Installing fonts. As I have switched my home PC to Rosa linux R8.1 I was finding it difficult to Install tamil fonts. I have searched in Internet then I got to know about installing language. Below is to check whether the language exists
urpmq -a fonts-ttf-tamil
After checking this Exists then install using the below command
Once done you will see the language enabled in the browser.
I have recently installed opensuse 42.2 gnome and making some changes. The changes were more about visual so I changed default theme as red arc theme and vivacious Icon theme then I changed color schemes in terminal and while searching I came accross this screenfetch.
Below is the error which comes when trying to mount NTFS file system with read and write access
The disk contains an unclean file system (0, 0).
Metadata kept in Windows cache, refused to mount.
Falling back to read-only mount because the NTFS partition is in an
unsafe state. Please resume and shutdown Windows fully (no hibernation
or fast restarting.)
After searching I got what to do
Run the below command
$ sudo ntfsfix /dev/sda8
Now mount the file system read and write access by below command
We can handle 96KHz 24-bit audio – but by default pulseaudio and alsa are configured for 44.1KHz 16-bit audio. At the 44.1/16-bit settings everything sounds great, but I figure if the quality can go higher and I’m not fussy about a bit of extra CPU usage then I may as well bump the quality settings up a bit to take full advantage. Here’s how to do it:
1 – Check current settings
To see what your current settings are, you can either run the following for the full details:
Or to cut to the sample rates directly use:
pacmd list-sinks | grep sample
Depending on how many sound devices are connected you should see something like this (in my example I have an Intel HDA internal soundcard as the first sink, and HDMI audio out as the second sink):
You might want to play an mp3 at this point and then look at the pulseaudio cpu usage via the top command so you can compare the current and HQ CPU usage as it might be a factor for you (i.e. depending on the spec of your PC, you might not want to sacrifice 10% or more CPU usage for higher quality audio – but seeing as you’re here and reading this I’ll assume that you are!). As a baseline, with the default settings pulseaudio uses around 2-3% CPU to play an mp3 on my Intel i7.
2 – Modify for high quality
Pulseaudio’s global settings are stored in /etc/pulse/daemon.conf, so edit that with your editor of choice and look for the following lines (the ‘resample-method’ line is above the default-sample-* lines btw – that is, they’re not all together as below):
These are the defaults, and are currently commented out (you can use either ; or # to indicate a comment). Uncomment each line and modify it to the below (or even better – just add an uncommented line below each one so you can still see the default values), so you end up with:
I get about 10% CPU usage on the pulseaudio daemon with 96KHz @ 24-bit audio with medium quality resampling as above, whilst using src-sinc-best-quality results in a rather significant 24% CPU usage just to play an mp3! ( It’s like having a Pentium 100MHz all over again! 😉 ) The ‘s24le’ means 24-bit samples, little-endian order, btw.
3 – Restart pulseaudio and check the settings
Now that’s done we need to restart the pulseaudio daemon (don’t run these as sudo – pulseaudio runs per user so just execute as normal!):
Once that’s done, you should be able to check the settings have taken by issuing:
It’s jumped up to 32-bit samples in this case, but that’s fine. Check the audio still works, and maybe have a look at the CPU usage when playing an mp3 via top again if you want to see how much this is affecting the CPU usage on your machine.
FiiO E17 Amp Check
Now that we’ve set higher quality default settings, if you’ve got some suitable hardware for them like the above-mentioned amp, plug that bad-boy in, switch it on – and it should validate that it’s getting the higher sample-rate and sample-depth:
So – does changing these audio setting make a definite, perceivable difference to the audio quality? The short answer is yes!
How much it changes the audio quality is going to depend on your hardware (amp, headphones etc.) as well as the actual audio file(s) you’re playing on it. I don’t have any 24-bit audio to try it out on, but I rip everything at the highest possible quality VBR mp3s using lame – so I’ve been testing on the first track of Alt-J’s album (intro) with some Sennheiser HD650’s – and my honest answer? I’d go as far as to say it’s a strong yes (html joke ;-)). The high-frequency elements in particular seem a little brighter with the 96KHz 24-bit sampling, at least when run through the FiiO E17. Now, this could be placebo – that is, you expect something to sound different/react in a particular way so you look for it and (internally, subconciously) try to convince yourself of it. But I’ve honestly swapped back and forth and listed to the same track a number of times during the writing of this article and I genuinely believe it’s sounding better with the HQ settings.
Unfortunately, the schoolboy error I’ve made here is that I’ve changed two (if not three!) things at once: I’ve enabled the higher sampling-rate and sample-size/depth, but I’ve also modified the resampling algorithm to one of higher quality at the same time – so figuring out what elements of the audio sound better because of the audio settings and which are from the resampling-method (because I’m not playing native 96KHz @ 24-bit audio files to test with) can’t really be done. To get any real results I’d need to strip back to default settings and modify each in turn, doing listening tests with notes as I go, which frankly isn’t all that high up there on my priority list of things to do because…
…I’m delighted with the audio quality after these changes! It really does sound great =D
Ideally, I’d prefer to have things set to 44.1KHz on each audio device EXCEPT the Fiio – that way, when I’m playing games there’s minimal CPU used on the audio side (which isn’t really going to benefit that much from upsampling and such, I’d imagine) and then have the HQ settings for the FiiO only so that when I plug it to listen to music I get the HQ audiophile kick. However, my understanding is that I’d need to go into the alsa side of things to configure device specific settings – and again, it’s not really a high priority at the moment.
I’d love to hear what difference (or lack thereof!) any of you find when changing over to HQ audio (asides from the bump in CPU usage) – and if you run or have played with different samplerate/re-sample settings for diff devices I’d be especially interested! Cheers!
Here you should put the results from running pacmd list-sinks and the information found on the line called sample spec. The first piece of information should be the sample format your card will work best with. For my case the output was
sample spec: s32le 2ch 44100Hz
thus my sample format is s32le. The float32ne is simply a generic method.
SECOND ISSUE: If you know that you will be using audio of a higher frequency (AKA something above 48KHz such as a Bluray HD audio track). Then you will likely want to change this area.
alternate-sample-rate = 48000
alternate-sample-rate = 96000
This will prevent resampling of audio that is @96KHz but it will create resampling of any audio @48KHz. As of right now I have no idea if more than 2 default sample frequencies/rates can be used. So you’re stuck with the choice of of which you want to avoid resampling. In my experience most sources even Bluray tracks seem to top out at 48kHz so I’d say leave it at 48000.
5) FOR MORE THAN JUST 2 SPEAKERS
change this accordingly (make sure to remove the “;” before each setting)
3 for 2.1 aka 2 speakers and a sub and (While 3 for 2.1 is valid and supported most 2.1 speakers don’t work properly with such a setting so you should tinker with that to figure out what works best for you)so 6 would equal 5.1 and 8 would equal 7.1 (you’ll have to add the locational name of each speaker you have in the channel map to make it work properly)
(a generic 5.1 setup edit to your needs)
SIDE NOTE: You might want to experiment with swapping subwoofer with lfe to see which one sounds best for you and your specific sound hardware. But make sure your hardware can handle the lfe as it is noted on occasion the lfe can destroy speakers or subwoofers.
6) IF YOU WANT TO MESS WITH LATENCY
Lower these numbers as much as you can (at 0 pulse will no longer work and ALSA will be used instead) but be careful they may introduce problems and flash will likely become a problem or may not work at all.
Apparently I goofed up here…though this is purely subjective to each person; the recommended thing to do is set a higher fragment number and size. For me I personally have had better results (lower CPU usage) with as few fragments as pulse will allow me (in my case it was 2) and as large a fragment size as I felt necessary (I’m at 400 right now). The changes here are really up to you. This section has been a source of problems as there’s much misinformation out on the net. The safest bet outside of default is to use more fragments and a fragment size of 50 or greater. This however is a generic method. If you want to know how to figure out the specific for your configuration setup for this part of audio you’ll need to be ready to do some math and have a read from the steps laid out by these guys at the mint forum: http://forums.linuxmint.com/viewtopic.php?f=42&t=44862
One note is that you should use your info from running pacmd list-sinks and then follow steps 2-3 in the mint guide.
6.1) IF YOU WANT TO TWEAK THE PRIORITY TO INCREASE QUALITY
Find “nice-level” and change it from the default -11 to a high negative (remember -20 is max and not recommended). Remember as usual to remove the ; before “nice-level”.
nice-level = -15#this setting increases the frequency/share of CPU time that Pulseaudio has access to
SIDE NOTE: Not confirmed by anyone else yet but it seems changing the priority has a negative effect on KDE’s system sounds so they’ll cause a second or two of glitching/blocky/ugly sound if something else is using audio. No idea why but the priority change might negatively effect sounds from DEs such as Gnome, KDE. Doesn’t seem to cause problems with other applications using sound though. UPDATE: Skype is another thing that doesn’t like the nice level or latency changes. If you make changes similar to this then make sure not have anything trying to use the sound server at the same time as Skype as it’s again one program that just doesn’t play nice with others. It works perfectly fine if nothing else tries to use the sound card so don’t worry there.
It should be noted that this will negate any of the changes made prior to this point as this is a lazy way of getting rid of pulseaudio without having to actually uninstall pulse audio. It works by simply causing pulse not to load as this is an incorrect setting. Thus, pulse should no longer show up in your process list or return a follow from running
ps -A | grep"pulse"
. This means that ALSA will be in charge of everything and configuring it is a whole separate topic. However, it seems most people believe that one does not have to configure ALSA at all as it should “just work” by default so you shouldn’t need to do any config work with it.
An alternative and likely better way to disable Pulseaudio is to change the following in the /etc/pulse/client.conf file:
; autospawn = yes
autospawn = no
7) exit and save changes
8) Next you need to kill pulse so the old config settings are no longer in effect
9) Then you should check that your config is valid
10) If it returns nothing then your config is fine and you can restart pulse with your changes implemented
NOTE: These changes may increase CPU load and the sample format may not be supported by the audio hardware on your mobo or card. Lower the number from 32 to 24 to 16 until it works. Otherwise use the original default setting. It should be noted that using a float setting will always result in lower CPU load and better sound quality.
In fedora they have grouped packages according the needs like for an example if a person is interested in he has to download a distribution or he has to get a fedora lab of his choice for example Audio Editing or he can go for some other distro which has Audio Editing tools and now he is also interested in security tools so he has to search all the tools and then he will install with audio editing tools. So to simplify this fedora provides group. So there are many groups users can get specialized tools in the groups.
Now type the below command to search the list of groups in fedora
$ dnf groups list
Now Install the needed group to your system by firing the below command
# dnf groups install “Security Lab”
Note: this command will not support the versions below fedora 21
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