linux-audio &laquo; WordPress.com Tag Feed http://en.wordpress.com/tag/linux-audio/ Feed of posts on WordPress.com tagged "linux-audio" Thu, 14 Nov 2024 05:01:29 +0000 http://en.wordpress.com/tags/ en <![CDATA[The Revival]]> https://linuxaudiolive.wordpress.com/2024/07/02/the-revival/ Tue, 02 Jul 2024 06:51:11 +0000 lal https://linuxaudiolive.wordpress.com/2024/07/02/the-revival/
Acoustic guitar-tuner on a paper copy sheet of a music score.

After almost a decade of silence, linux audio live is back with more than ever. More skills, more experience, and above all, more determined. Their stage gear has seen a tremendous upgrade: a year-2017 laptop powers a two-in-two-out audio interface. The current line-up counts with an electric guitar and eight midi pads and knobs.

Behind the scenes we can still see SooperLooper running, that beast of a live looping sampler! Guitarix, one of the new members, takes care of all guitar amplification needs. All audio is eventually channeled into jack_mixer, displaying and controlling every single sound going outside.

That roster has proven stable with less than 10 milliseconds latency and things are much easier than in the old times. The first post on linux audio live, in 2008, was a how-to set up real-time kernels, another post on the same topic followed in 2009. Today, real-time kernels work out of the box on all major distributions.

It is exciting to see what lal is going to bring on. There probably will be some visual redesigns, likely one or two new partners, and maybe even new colors and typefaces. Who knows? The table is set and everybody is waiting.

]]> <![CDATA[How do I prepare Manjaro KDE for audio production?]]> https://internetmarketingbusinessblog.com/2020/06/28/how-do-i-prepare-manjaro-kde-for-audio-production/ Sun, 28 Jun 2020 17:20:00 +0000 Jimmy Dagger https://internetmarketingbusinessblog.com/2020/06/28/how-do-i-prepare-manjaro-kde-for-audio-production/ How do I do that? I have a fresh Manjaro installation and we’re going to go for the process of configuring, the system and installing all necessary packages making sure it works good and then installing all the software that you might need for a music production, all the doors or the plugins.

And that sort of thing, and I’m going to show you where to get this and how to install it, how to later update it, etc. So you’ve just logged in for the first time into your new system and there’s a few things I want to do. The first thing might be a little bit strange to you, but the first thing I ever do is configure backups. So let’s open a program called time shift. Now it’s not installed, so we have to install time shift now.

Something I really like about Manjaro is that it has a drop-down terminal installed by default. So if you just press f12, you get this terminal, you can press shift, +, alt, +, arrow keys, right, arrow, key and down arrow key to make it bigger, and I usually like to have it much bigger. We can also make the font bigger if we just use ctrl + my mouse wheel, and this is going to be much easier to to follow.

I hope alrighty. So we need to install time shift, but first we need to update our package list so pac-man, which is the package manager s and Y. This will go into in update all the package information but of course we need to use sudo so sudo and I can now type exclamation points and that’s going to execute the previous command with prefix. So that’s going to execute pseudo pac-man s. Why? Now? I’m going to type in my temporary password, which is one two three four and let’s let it run and everything is up to date great now we can install time shift so I’m going to go sudo pac-man s, time shift.

This is going to install a package called time shift and all its dependencies proceed with installation. I just press ENTER and it’s default. Yes, it’s going to download some packages and time shift is installed. Now I can press f12 to hide the terminal. We can also close this hello dialogue. Now I can open my main menu and type time, and you see there is time shift a system restore utility. Now we need to give it our administration password that you’ve set during installation and during first run time shift, is going to ask you to configure how it’s going to do its job.

Now, if you have installed your system using ext4, filesystem you’re going to have to use our sink, which is the default option, this is simpler and but it takes longer to do. I have installed Manjaro using Manjaro architect and I’ve installed it on a btrfs filesystem. So I can use the very fast snapshot functionality of btrfs to create and restore these snapshots nearly instantly. So I’m going to select btrfs, you can also open help and read up about them.

Now we have to choose the file system, we’re going to store the backups on. We have one file system present in this in this laptop, it’s the main file system. So we don’t have to do anything now. We can configure a snapshot schedule by default. Time shift is going to do five daily backups, I’m going to disable that, because I want to be using time shift only to do manual snapshots when before upgrading my system, let’s go next this, let’s leave this enabled, and now we can include the user home directory.

What this means is everything in our home user directory will also be included in the snapshot. That means our documents. I want to use time shift only to store the system state, so I’m going to leave this off, but if you want to also include your documents, projects and everything, you can click this and it will back up everything, let’s press next and the setup is complete. I can now click finish: let’s create our first snapshot.

What you got to do is just click create now on btrfs. This takes no time at all. If you want to delete a snapshot, just use this delete button. If you were to roll back to that snapshot, we just press restore it’s as simple as that. We can also browse the selected snapshot files to retreat, something that was previously on our disk. But it’s not isn’t there anymore. Ok, this article is not about time shift.

Let’s get going now, I want to make sure octopi is installed and you can see it is, but in some previous versions of Manjaro it was not installed by default. So if you would like to install octopi, you would go to your terminal window and we can clear it and you can type sudo pac-man s. Octo P, give it your administration password, and indeed there is an update to be installed for that program. Let’s go ahead and do that all right.

What actually we want to do is update the system in its entirety, so I’m going to clear it the terminal and I’m going to run sudo pac-man, big s, small Y and small. U this command will update our system. Basically check if there is anything to update and it will update everything. This is an important thing on Manjaro, which is based on arch linux, as arch. Linux is a rolling release distribution.

That means there are no frozen states like with Ubuntu’s. You know different trees like 1404 or 1909 or who knows what 1909 oh, and then you update packages for the versions supported by this version of the system. No on arch based Linux distributions, like Manjaro, you are always on the bleeding edge, so you often will be required to update all your system packages now there is not much to update, because I have actually updated it during installation.

Another tool we’re going to need to install extra software is called yay yay, and it’s going to give us access to so called arch user repository, which is a collection of user created. Pk builds. These are scripts that get sources or binary packages from the internet and either build these sources or build packages from the binary files and install them to your system. There is pretty much everything you could ever imagine or want in the arch user repository, so we’re going to need that for some really bleeding-edge versions of our software or plugins that are not in the usual repositories.

So let’s go sudo pac-man s yay! Now you can see that I already had ya installed and that’s because I have chosen to have it installed during installation when I used Manjaro architect. But if you do a regular installation for the graphical installation program, you’re not going to have that option. So you will need to install yourself. The next thing we need to take care of is the jack on your server.

It’s pretty much the workhorse that’s going to hold all our audio work together. So, let’s just see if we have Jack installed, let’s go which Jack D and you can see we have Jack installed, but we want to make sure it’s the correct version. So I’m going to just make sure that Jack two is installed. I don’t use Jack one because Jack one is not multi-threaded and modern computers have at least two cores.

So it’s a shame not to use them for your DSP. From what I see here, Jack one doesn’t have any cool features at all: let’s go sudo pac-man s Jack and now you can see we have jacked installed so by default. Manjaro has supplied it with Jack 1, but we want Jack to do you want to remove Jack. I do so. We want to replace Jack 1 with Jack. Yes now. This is a very important step because if I have gone and installed other software depending on the jack replacing that package later would be quite problematic, because I would have to uninstall all the software that needs Jack audio server.

Then I could remove the jack 1 a package and then install Jack, and I would have to reinstall again all the software installed. On top of that now we are sure we are running Jack, so we can go ahead and install other things and the other thing I’m going to install is cadence sudu pac-man s. Cadence cadence is basically a control panel for production, audio work. Okay, we have cadence installed, let’s hide our terminal and let’s find it cadets.

Yes, it’s a jack toolbox, let’s open cadence and we can see. We have a yellow sign here, which says we have a non real-time kernel and with the user is an audio group. We’re going to check that soon, let’s see what we can do in the configuration the driver, we want to use alsa and yes, this is the analog interface, okay, fine, and we need to use real-time priority. Indeed, to up our performance, we can move the real-time priority up and now, if we try to do that, the jack server is started, but I’m going to stop it for now and first make sure we have other things configured properly.

So a one important thing for using Jack and Jack dependent applications like our door is to make sure that your user is allowed to use real-time scheduling in the system, and this is done with the use of so called audio group. So we can type in a command called groups and that’s going to show us what user groups our user is in and we can see it is actually in the audio group. But we need to make sure that the only group is properly configured so that it has the privilege to use the real-time scheduling that the jack server needs to do that, I’m going to go to a directory called etc’ security.

I’m going to list the files in there you can see there is limits, conf and limits dot D. This is a monolithic configuration file and this is a directory containing smaller files, which can be which are concatenated to produce the complete configuration I’m going to go into the limits, dot, d directory and list. What’s there you can see, there is something, let’s, let’s type less and ten and see what it is: their users mem, lock, 2024.

Ok, there is nothing here. Let’s go up, let’s now type nano limits, conf and I need to prefix that command with sudo. So this is run as administrator’. Nano is a text editor and limits. Conf is the file we want to edit and that’s the contents of the file. Now, let’s see if the audio group is mentioned here, let’s press ctrl W, which stands for where and now I can type audio audio is not found. That means our audio group is created in this system and our user has been added to it, but it doesn’t really have the privileges that we need.

So let’s go down to the end of the file and specify the privileges for the audio group. Let’s type a hash which starts a commented line and this is going to be ignored and now I’m going to type at audio, which means the name of the group user group tab. I’m going to use two tabs just to aligned with what is above. The type is nothing. The item is RT prior, which means real-time priority. Now I’m going to allow it 95 and, let’s also add another line, let’s call it audio again type tab, and this is mem lock unlimited.

This is another privilege for the jack server to use more memory in your system. I’ve taken this text from a website, Jack audio org, and there is a frequently asked questions section where you can find exactly how to configure this so yeah. You need to look it up and you probably will need to look it up now. We need to save the configuration file, I’m going to press ctrl o for out or right out limits.

Our conf enter. It wrote 54 lines great call. Now I’m going to press control X to exit, we can verify that the file was changed by typing less limits conf, and this is a file viewer. We can press page down, and indeed there is our audio group privileges specified, let’s press Q to quit this less problem now for this to take effect, we need to log out and log back in. So let’s do that. Okay, I’m logging! Back in now, we have cadence opened and can start all right now.

If there was no audio group in our system, we can create it by typing. Sudo group add audio. Now we need to type in our administration password. It’s going to tell us that the audio group already exists, but if it wouldn’t exist, we would have created it just now. Now we would have to also add our user to the audio group, and we can do this with the command usermod’. A big gene which means add to a group audio and now we can type our username alpha, but you can also type dollar sign and with big letters, user and this text will be substituted with whatever is the username.

Of course, I need to prefixed s with sudo or afterwards I’m studio and exclamation points, and this will execute the command. Now I have now. I should have done this before configuring, the audio group, because otherwise, that configuration would have not made any sense and then log out log back in and it would take effect to verify, let’s type in groups and indeed we are in the audio group. We can make it a bit easier by typing groups, pipe grep, which is a program to search for text audio, and it highlights the audio group in red for us to make it easier to spot.

Our user is in the audio group. Our audio group is configured so we should be able to run our tour just fine, let’s install it now. Maybe I’m going to use octopi to do so. So far I’ve been installing everything using the terminal, but you might prefer to use the graphical user interface. I’m going to type in our tour right, click install and it asks us to install optional packages. Xj do is an X, Jack article viewer, it’s a very useful viewer for working with film, soundtracks or editing, or making sound effects for pilot footage or whatever I’m going to install both of these.

Now, let’s commit these changes or apply these changes. Now I need to, of course, give my administration password and it’s going to install our door for us and Artur is installed, so let’s run it our door. Welcome to our door, I’m going to put everything in default places and I’m going to change this later. Let’s scan for plugins there’s not going to be much plugins because we haven’t installed any: let’s create an empty session hard or test my favourite name.

Let’s change the audio system to Jack, which is already running because we have started it with cadence and we can connect to it now. If we wouldn’t have the audio group and the audio group privileges properly configured, our door would not be able to create this session. It would give us an error and say that it cannot open the the session file or whatever it’s a misleading error message and what it really means is that it couldn’t obtain the real-time scheduling privileges.

Now we can pin the ardor icon to the task manager. It will let us open it up more quickly. You can also right click to start a new instance. If you want to open two other instances for some reason, we can create a new track control-shift-n, let’s make a stereo audio track, and this is already capturing audio from the built-in microphone huh. Let’s see if it works, it does hmm alrighty. It works. Now.

Let’s install more software, I’m going to save this. Let’s go back to octopi and now do you remember. I have told you we installed this program called yay. Now. If we go to tools, options and aur, you can see. We have yay selected as a as a program to interface with the arch user repository. You may have this dis elected so make sure you select yeah here. You can also enable a you are voting if you created an account on our aur Arch Linux at org.

I know I have so now I can vote on packages. You can select, no confirm and no edit to make it easier for you to go through the installation process. This will disable the option to change anything in the pay, Pete a build script, but it will also not ask you as many questions and it just going to be smoother. Let’s apply these settings and now I’m going to install this infusion first thing: I have to click on this alien face to enable the eye tool so we’re going to search the arch user repository instead of the regular arch linux or Manjaro linux packages.

So i’m going to type in Zen and press enter and we have Zen fusion here, let’s right-click and you can see. I have already voted on this package. Let’s install it and now octa-pie opens up an interminable ater and runs the high command with parameters of our package. It’s going to ask us a password, I’m going to give it the password, and you see it shows question proceed with installation and automatically answered yes, because we have selected the know confirm option.

This is right now downloading all the source code for s infusion and it’s going to compile it on the spot and install the package for us, so we can use it. This is going to take a short while, because the infusion is quite a big program now, as this installation is going on, I’m going to tell you something about pulseaudio, pulseaudio and jack are a cause of many headaches for many people using linux for audio production, because Both pulseaudio and jack are sound servers and they tend to not cooperate very well, but there is a very easy way to deal with that, and that is just to install a jack module for pools audio.

Then Posada will be able to use Jack as its audio backend. Instead of trying to use your sound card directly, two nights are fighting for the attention of Princess Elsa, the nimble Jack and the well-known police audio. But for the kingdom to blossom, we need Jack to pass the letters between Elsa and pull so do as any other way around will cause a disruption. What did I just make up unless all of that can be handled by pipe wire, but that’s a topic for another article? That means you will be able to play articles from your web browser and run order at the same time without having to stop or the other, and you can also then record articles from the web browser into your other session, because pulseaudio will expose its audio ports To the jack server and Kadence makes this easier you can see.

We have this post audio tab right here and we have information of the post of your server. You can see that post audio is started and bridged to Jack. That’s good news, because that means we already have pulls out you configured to talk to Jack. You can go to tools open katia, which is a Jack patch Bay, and we can see what is happening in our system right now. This is a graph of Jack clients and connections.

The system nodes show the built-in sound card that we are using for our Jack server, and you see we have Poots audio jack sync, which is the post audio output. So whatever you play in your web browser, for example, is going to feed it there. It’s audio here and right into our speakers, and here is post audio jack source, whatever you’re going to, if you’re going to talk to someone using a web application in your browser.

This is going to get the sound from your microphone, and here is the stereo microphone built into the laptop, but with the power of Jack, you can reroute that wherever you want, for example, you can feed output of your other session to pull Solio, so we can, For example, process your voice with our door and make people hear you with your voice changed or with reverb added or whatever I’m going to reset that to the previous st.

. Do not leave it in a mess. So Katia is a tool shipped with cadence. There is another tool called Claudia, which is a front end for laddish session mannered manager. There are also level meters. This one shows the microphone input and this is an out meter and I think it’s working, because we have this thing here. Yes, let’s see how the installation is going and we’re still cloning. Now we are copying the downloading the source files for EM Robby’s ESTs framework, which is the GUI framework that Mark Mercury have created for this infusion.

Normally this is much faster, but I’m recording the article and that slows everything down. Another thing we can talk about is using real time. Kernels now with Manjaro. Installing a new kernel is very, very easy. You just open your main menu and type kernel, and there is this tool for installing different kernels, and here are all the available kernels you can see. We are currently running kernel version for 5.

4 point 18 1, which is a long time supported kernel, it’s installed and it’s running, but there is a newer kernel, but there is a different kernel and it’s a real-time kernel. So a real-time kernel might give us better performance for Jack audio server. However, I have not found a significant change difference in my workload, but if we were to install it, we will just click here. Click yes give our password and we can show details and it’s going to download all the packages and install the real-time kernel for us, of course, to use this new kernel, we need to reboot our machine and you can verify what kernel are you running either by Opening this Manjaro kernel manager, or by opening a terminal hit f12 for example, and type a command called, you name a, and this tells us we’re running Linux.

This is the name of the computer, and this is the version of the kernel. This is a pre-emptive kernel, so it allows real-time scheduling, but it’s not like super full support and our new kernel is installed. You can see it’s now installed. If we reboot our computer, you will have a menu where we can choose which kernel we want to boot. Once we’re done with installing all the packages, I will reboot and we will choose this real-time kernel and once we confirm that it works, then we will remove the other one.

So we don’t have this option and the boot process will be simpler. Let’s check back and we’ll tale cloning by the way, a very important command on Linux is sudo, make coffee just joking. Okay. Now we have to give our password. One thing I don’t like about yeh is that it can timeout with the password. I need to install it again. Thankfully, it caches the results. So it’s not a big problem and we have the infusion installed.

Let’s try it in our door, I’m going to press Shift key to open up the side terminal by the way. I think we should change the font size of our door now there it is yeah that looks much better. Not all phones will be updated immediately, though you should rest at our order for this to take full effect and I’m going to press ctrl shift and and add a new track. Let’s make it a me track and you can see we don’t have as in fusion on this list.

What we have to do is go to edit preferences plugins and go scan for plugins, and you can see we have it’s detecting, it’s helped. It has detected zeenat’s of effects so now, if I go to control-shift-n and add a new MIDI track, I have the amounts of the facts on the list. I’m going to pick lv2 version. Let’s call this Zen fusion for some reason it didn’t have to plug in here: okay, let’s do it CIN and some effects.

Now it’s failed. What the okay all right! So now we can control shift and create a new MIDI track. Okay, let’s press control shift n to create a new MIDI track and I’m going to choose Xin fusion VST. Let’s call this Xin fusion, it’s called dance of effects, but with the new interface it is as in fusion and there we have it and it works. I’m going to install more software that I use and I need let’s go, install surge and there is search synthesizer.

We can install the binary version, it’s going to be faster because it’s not going to have to compile everything. So if you read the output, you can see that it’s downloaded a debian package right here. It’s extracted it and it’s created an arch package from that. Now it’s compressing this package, so it can be installed. So arch user repository is using various means to pull in different software, so it can be installed on Arch Linux based operating systems like Manjaro.

Now it’s installing the package and it’s installed. If we now go into edit preferences and scan for plugins, you can see it has found a search plug-in. You can create a new MIDI track, call it Serge and we can select search from the drop-down menu and, of course it doesn’t work. Hey it’s alpha and it’s eight days later I have actually used this laptop with our door 6 and that we installed during this article that you’ll see and I recorded a gig and everything went perfectly and right now.

I’ve just went back to check this order. 5 problem with all be two plugins and I removed order six and installed our door five again and it works it works. So I think that must have been some error in the arch unofficial order build and that it’s already fixed. Okay, let’s install a bunch of plugins that I use all the time I use called plugins. They are very important. I also use LSP plugins tap. Plugins may be an easier way to do.

This would be to go to the raw audio group of packages, and here we have a lot of stuff that you might want to install, for example, audacity yep golf caps, also Carla the plug-in host. This show plug-in framework, dragonfly reverb. Of course AB you meet er. Yes, drum gizmo, oh yes, EQ tank. You certainly G on kik. Oh yes, guitarist for sure helm, yeah, IRL v2. It also might want to install Jack capture or something else.

I don’t like Linux sampler at all liquid sfz I’ve heard it’s a nice plugin LSP plugins yeah MDA lv2 yeah. These are some good plugins, though I had problems with them. Ninjas too, is a nice drum loop, slicer noise repellant is an excellent realtime de noising plugin. Actually two plugins one is using neural networks can looking through anything else I might need or want sample v1 is a very useful sampler set b3 is an excellent hammond organ emulator Sherlock lv2 is a great plugin for reading.

What’s going on in your MIDI stream, sonic visualizer is a very useful program for well visualizing audio file contents, as WH plugins are used to Tampa plugins sure wall shaper. Oh yes, love spectrum 2 X, 42 plugins are very useful. These are excellent. Audio metering, plugins zam plugins, are great. Zita audio jack bridge is very nice. It’s a tool for bridging, also applications to Jack. A Zeta ng bridge is a network jack application to send Jack audio over network.

I’ve made a article about that and there’s 8 RF 1, which is the very well sounding reverb and now to install them. I right-click somewhere and then go install. It asks me to install Jack. I don’t want to because that’s optional and I also have Jack already project. Em is a music visualizer. I don’t need a visualizer plugin and it should now list all the packages I want to install. Do we have everything I think we have everything we need.

Let’s do it, of course you need to give it your password, and now we just wait in the meantime, I’m going to close this our door session, so the user interface can reload with the changed font size. Now, we’ve mostly installed software that is present in the regular Manjaro or arch linux repositories, but we can also fund search plugins that are in the arch user repository. Sometimes these are newer versions of the plugins.

I’m always amazed how fast Arch Linux packaging manager works on Debian it’s like 10 times slower. I got used to how slow it was. It’s already installed. Alright. Now we can just go edit preferences plugins scan for plugins. Now that’s going to take a while, okay yeah now this is unlocked. I can enable the aur search. What is there that is not in the usual repositories, I’m going to search for LV, Oh bitch, rot bit.

Rot are very useful plugins that are unfortunately, not very popular. We can install bit rot plugins. Let’s do that. It’s going to download the sources of of github and build them for us, these are very small, so it shouldn’t take long. I very like the bitch rot, repeat plug-in, which is very good for on creating glitchy effects, and you can very well automate it in order. I like to use it on drum lines, especially so you can have like, oh in Nevada, studio plugins in verse.

Three planes will be yay in Nevada are also some great plugins. Now I need to give my password. Okay bit. Rod is installed. We can install the invader plugins, it has 44 votes. People like these. Let’s see how order is going. Okay, it scanned everything. It has all the new plugins detected. Now we have a lot of plugins. Let’s go EQ 10-q. I use these a lot. So I’m going to add these two to my favorites.

I think there’s something wrong with our disorder installation. It can’t load any lv2 plugins. That’s no good, usually it just works so kind of sucks and I really wished something like that: wouldn’t go wrong when I’m making a article, but we have our in Venice to do plugins installed. You know what I’m going to do. I’m going to install our door! 6.0 and hope it’s going to work a bit better, so note that I’m searching in a you are, and we have our door get let’s right-click and install.

It would like to ask us to remove our door first, because these two are conflicting. If I were to edit the PK build file script, we could install both alongside, but let’s not do that, so I’m going to just remove our door, it’s already removed. Now, let’s go back to the aur search and I’m going to install our door, get you can see that our our door desktop or our icon here is no longer pointing to anything I’m going to unpin it because our door 6 is probably going to add a Different icon, we can also run Carla the nonlinear plugin host.

We can add a plugin, let’s see if the lv2 plugins work in Carla, let’s go with in Nevada. I can go with special filters and, let’s just enable lv2 plugins, nothing else in Fatah. Early reflections, reverb loaded and here is the user interface. I really like this early reflection, reverb plug-in, because you know it has this three-dimensional visualization of a room and you can place your source and your listener, and you can see the impulse response being generated on the fly and visualized.

I’ve used this a lot in the Lance perversion and lmms: okay, so lv2 plugins work in Carla, so that some problem with order. So we are not installing official order builds. So we can’t complain to order developers because the official builds probably work. Let’s open Serge lv2 here and it works, let’s make it bigger yep, just like it should alright, so Artur 6 is installed, let’s see if it works out or 6 of ascared configuration from our tour 5.

Would you like to these fans to be copied and used for our door, 6, yeah, sure and congratulations file will cover you now. You now can restart our door and okay. I don’t know why this get version. Has this weird icon welcome to this pre-release build of order? Six point: zero: three: zero! Three. 504! Yes, I know don’t use it for production, let’s create a new session, our door, six test. The main thing I want to know, if is if this version of our door, is going to work with LV two plugins, because that’s quite important, and I have no idea why the order 5 version didn’t work and, let’s see if our r6 works with all v2 Plugins because our door 5 didn’t want to control-shift-n, let’s create a new MIDI track and insert maybe sentence of effects.

Let’s call this Xena, and here it is, I’m not sure if this is the LV to plug in or the VST one. Let’s add a new track. Let’s go to me track: let’s make a no plugin at all and let’s insert the plug in manually, so plug-in manager, and here we should be able to filter by type so LV. Okay, let’s show instruments only show all formats. I want just LV okay, so we have LV two instruments: let’s try G on kick okay, so LV two plugins work in this order.

Six build that’s good. It took a while to install because this laptop is not the most powerful one. It just has 4 cores, and but it did it alright, so we have a working workstation to do things now. If we wouldn’t have the pulse out your jack module pulsar, they wouldn’t cooperate with jacks, so let’s try and find it in the package manager. In case you need to install it, we can also install vcv rack.

So in aur we have vcv. Rack can install the binary package, so you don’t have to build it from source. Ok, that works, and you can see we have VCB rack here. We can change the audio backends to Jack and we can make it feed into the system output, but now we can intercept that you can clean these and use these outputs here and now can record. Don’t pay attention to the X runs. This laptop is under heavy load.

No, we have recorded audio from vcv rack into order now. The very last thing we can do to try and improve our jack performance is to install a real-time kernel so away like I said before, you can go here, type kernel and we’re going to have this thing opened now we have installed the real-time kernel, but we Are still running the non real-time kernel, so I will now reboot this machine. I like to just press f12 and type reboot, oh by the way before we do, that we can create a time shift snapshot and we can do it in two ways.

I’m going to show you the graphical user interface way, so you’re open up time shift. We give it already you administrator password or root password, and we can click here to create a new snapshot, but I also want to show you how to do the same from the comment line. So I’ve got sudo time shift and if you run this command, it’s going to show you all the help text to tell you what you can do with time shift.

Basically we’re going to use the create option with comments. So let me clear the terminal and I’m going to go su time image if create, and then comments and now I’m going to type after installing all this software and I got press ENTER and it’s read a bit ers snapshot now. If I run time shift note time shift list, oh sorry, I need to use sudo, you can see. We have two snapshots. This was our first one, and this is the second one I’m going to close this terminal and I’m going to open the graphical user interface and you can see we have the same thing here and we can double click here to type comment, and this is going To be our initial snap shot snapshot, so it’s nice and we know what is what and why these are here.

You can see that our size has changed to four point: six gigabytes, so there’s quite a lot of things changed in our in our system partition or file system alrighty. I am going to now reboot the computer, so I’m going to go and you can do it from the main menu here leave and press restart or be cool like I am and just type reboot. What I’m going to do now is press shift to open the grub menu, but it doesn’t okay.

I think we need to change the configuration of grub. So let’s go sudo nano et Cie default grub, that’s a grub! Let’s give it the password, and here it is grab a timeout style uncomment. This line, if you want unable to save default function, come in the following line and set group default to saved okay Group default is set to saved, but this was disabled and I’m going to allow the generation of recovering boot modes because that’s useful, sometimes, okay, I’m Going to save this file and go update grub now, I need to, of course give it sudo, and it should use this to remake the configuration files.

Now if we reboot, I think we should be good if I’m going to smash their shifts key anyway. Get me that group menu group Group group group, oh yeah, we have the menu now we’ve got the group menu and the absolutely amazing thing about time shift and btrfs is that now we have arch linux snaps and you can see, we have the snapshots that we’ve Created so we can basically boot right into an older version of our operating system.

If we messed something up, we can just select a different snapshot right here. For example, if we removed a working kernel – and we don’t have anything to boot – we can just go here and boot an old version, and this is something really amazing – about btrfs and timeshift snapshots. Now we can go for Advanced Options for men, general Linux, and here we can choose the kernel. So man Gerald Linux, kernel 5.

4 is the default one and we’re going to go for 5.4 13 real time and I’m going to press ENTER and boot this one. Now, every time I boot into btrfs, I get this error and really doesn’t mean anything. You don’t have to press any keys. It’s going to disappear really soon. Okay, so we have logged in let’s open the kernel, dialogue see what we’ve got and yes, we are running the real-time kernel. Well, I hope you found this article to be useful thanks for reading and all the good luck with installing your own Manjaro system and configuring.

It for audio work. It’s not that hard. Many things that I’ve shown you are kinda, like double checking that it’s correctly installed. Now your the mileage mayor of may vary. You know. Every version of the mange road is so that you can download is going to be slightly different and also, if you install using the Manjaro architect or the graphical program, it’s going to give you different results. But overall, it’s really not that bad yeah.

So thanks for reading, I hope you’ve learned something and also big thanks to all fun people who are supporting my work on beat John and litter. If you would like to join them and help keep this show going, please go to patreon.Com/scishow and fur now go install Manjaro and make some music hey I’d like to apologize, because this article has taken away too long to finish alright. So I’d like to apologize, because this article has taken way too long to finish, however, midway through the editing, all the article editing in open source hell broke loose and I had to abandon olive altogether because it became so unstable.

I was unable to proceed and I had to go back to kdenlive, which also isn’t working great and it’s just crashing and hanging, and so it was real hell to finish this article, but I thought it yeah, so I hope you’ll enjoy it. I really hope it’s going to be worth it. Ok, bye,

Videos are truly an awesome way to get the point across. Any type of content from your business is important!

 

]]> <![CDATA[How to fix Debian/Slax Linux playing no sound on laptops]]> https://suryanshpradhan.wordpress.com/2019/12/11/how-to-fix-debian-slax-linux-playing-no-sound-on-laptops/ Wed, 11 Dec 2019 22:51:17 +0000 Suryansh https://suryanshpradhan.wordpress.com/2019/12/11/how-to-fix-debian-slax-linux-playing-no-sound-on-laptops/ I encountered this issue while trying to boot a Slax Linux (https://www.slax.org/) image on a laptop. I’ve read that this issue sometimes occurs on Debian as well, which is plausible since Slax is based off Debian now. I haven’t tested the fix on Debian yet, but it should work in theory since they’re related.

The issue in itself is the user not being able to play audio and encountering errors like these:

VLC fails to play audio

ALSA Basic Audio Test fails

speaker-test fails

 

 

 

 

 

 

 

 

 

There are multiple ways to fix this problem, some more simpler than the other. The crux of the matter is Linux detecting multiple audio devices and using one that’s not actually in use (example: using the missing HDMI audio output as default instead of the usuable analog). We fix it by changing the default audio card.

Method 1: setting the ALSA configuration file at /etc/asound.conf or ~/.asoundrc

ALSA stands for Advanced Linux Sound Architecture, the framework that connects the Linux kernel with sound card drivers. We need to configure ALSA so that it uses the correct sound card for audio playback.

The /etc/ directory is used to hold configuration files. Our ALSA config file needs to be named asound.conf. We need to find the id numbers of our connected sound cards and create the asound.conf configuration file to set our desired device as the default. From the ALSA project wiki [https://www.alsa-project.org/wiki/Setting_the_default_device and https://www.alsa-project.org/wiki/Asoundrc], we can see the ids of the connected sound cards with the command:

cat /proc/asound/cards

cat /proc/asound/cards

We can also list all sound cards with aplay -l

aplay -l

We can then use any text editor to make the asound.conf config file and put the following text in it:

pcm.!default {
type hw
card 1
}
ctl.!default {
type hw
card 1
}

OR

defaults.pcm.card 1
defaults.ctl.card 1

Editing asound.conf

I’m using the id 1 to set the Intel PCH analog sound card as the default instead of the HDMI card which clearly doesn’t work since it’s not connected to an HDMI output. Replace 1 with the id of whichever card you want to use. Save the file and exit. The audio should start working now.

Alternatively, instead of making /etc/asound.conf you can create .asoundrc in your home directory instead. This is useful if only want to make changes for your own user and not system-wide (which is what happens with asound.conf).

Method 2: Using PulseAudio

PulseAudio is a sound server that works with ALSA and routes audio streams. It comes pre-installed with most Linux distributions and desktop environments. However, Slax itself doesn’t come with PulseAudio. We’ll need an internet connection to download PulseAudio on Slax. Use apt to install it:

apt-get install pulseaudio

[use sudo apt-get install pulseaudio if you’re not root/superuser]

Start the PulseAudio server with the command:

pulseaudio --start

Now, use the pactl command to set the correct sound card as the audio source for PulseAudio. The PulseAudio server system uses sinks to get and play audio from sound sources

pactl list short sinks       #list all sinks
pactl list short sources     #list all sources
pactl set-default-sink 0     #set default sink, replace 0 with your desired sink id
pactl set-default-source 1   #set default source, replace 1 with your desired source id

Match the correct sound sink to sound source with pactl

You can also alternatively use pacmd for an interactive shell to configure PulseAudio. The commands are essentially the same.

]]> <![CDATA[Setting up analog surround sound on Ubuntu Linux with a 3 3.5mm capable sound card]]> https://dennismungai.wordpress.com/2018/02/06/setting-up-analog-surround-sound-on-ubuntu-linux-with-a-3-3-5mm-capable-sound-card/ Tue, 06 Feb 2018 22:45:22 +0000 Brainiarc7 https://dennismungai.wordpress.com/2018/02/06/setting-up-analog-surround-sound-on-ubuntu-linux-with-a-3-3-5mm-capable-sound-card/
A while back, I received the Logitech Z506 Speaker system, and with Windows, setting it up was a pretty plug and play experience. On Linux, however, its’ a wholly different ballgame. For one, there’s no Realtek HD Audio control panel here, so what gives? How do you around this problem?

Introducing the tools of the trade:

You’ll want to use a tool such as hdajackretask , pavucontrol and pavumeter for the pin re-assignments and audio output monitoring afterwards respectively. The tools are installed by running:

sudo apt-get install alsa-tools-gui pavumeter pavucontrol

When done, launch the tool with administrative privileges as shown:

gksudo hdajackretask

From here, you’ll then need to re-assign each required pin. Note that this tool, depending on your sound card, will most likely detect them by the color panel layout (see the back of your card and confirm if its’ pins are color coded) or by the jack designator.

Either way, when you’re done and you select “Apply”, you’ll need to reboot and the settings will apply on the next startup.

Before you reboot, confirm that pulseaudio is configured to utilize the channel layout as desired.

Of note is that for /etc/pulse/daemon.conf , the following changes must be made (with your preferred text editor):

(a). For 5.1 channel sound, set: default-sample-channels = 6

(b). Ensure that enable-lfe-remixing is set to yes.

(c). The default channel map option for 5.1 audio should be set as:

front-left,front-right,lfe,front-center,rear-left,rear-right

How the tool works:

The tool generates a firmware patch (under /lib/firmware/hda-jack-retask.fw ) entry that’s also called up by a module configuration file (under /etc/modprobe.d/hda-jack-retask.conf or similar) , whose settings are applied on every boot. That’s what the “boot override” option does, overriding the sound card’s pin assignments on every boot. To undo this in the case the configuration is no longer needed, just delete both files after purging hdajackretask.

An example:

To get the Clevo P751DM2-G‘s Audio jacks to work with the Logitech Z506 surround sound speaker system that uses three 3.5mm jacks as input for 5.1 surround sound audio, I had to override the pins as shown in the generated configuration file below (confirm with the screen shots attached at the bottom for my use case, your mileage may vary depending on your exact sound card):

(a). Contents of /lib/firmware/hda-jack-retask.fw after setup:

[codec]
0x10ec0899 0x15587504 0

[pincfg]
0x11 0x4004d000
0x12 0x90a60140
0x14 0x90170110
0x15 0x411111f0
0x16 0x411111f0
0x17 0x01014010
0x18 0x01014011
0x19 0x411111f0
0x1a 0x01014012
0x1b 0x411111f0
0x1c 0x411111f0
0x1d 0x40350d29
0x1e 0x01441120
0x1f 0x411111f0

(b). Contents of the /etc/modprobe.d/hda-jack-retask.conf file after setup:

# This file was added by the program 'hda-jack-retask'.
# If you want to revert the changes made by this program, you can simply erase this file and reboot your computer.
options snd-hda-intel patch=hda-jack-retask.fw,hda-jack-retask.fw,hda-jack-retask.fw,hda-jack-retask.fw

Then rebooted the system. Confirming the successful override by running grep on dmesg on boot:

dmesg | grep hda-jack-retask

Output:

[    5.183912] snd_hda_intel 0000:00:1f.3: Applying patch firmware 'hda-jack-retask.fw'
[    5.184524] snd_hda_intel 0000:01:00.1: Applying patch firmware 'hda-jack-retask.fw'

Confirming the 3.5mm audio jack connections to the sound card on the laptop/motherboard setup:

On the rear of the Logitech system, all the I/Os are color coded. In my case, I swapped the GREEN line with the YELLOW line such that the GREEN line feed would correspond to the Center/LFE feed, as it does on Windows under the Realtek HD Audio manager panel. Then, on the computer, I connected the feeds in the order, top to bottom: Yellow, Green then Black at the very end.

Final step after reboot to use the new setup:

Use pavucontrol (search for it in the app launcher or launch from terminal) and under the configuration tab, select the "Analog Surround 5.1 Output" profile. This is important, because apps won’t use your speaker layout UNTIL this is selected.

When done, you can verify your setup (as shown below) with the sound settings applet on Ubuntu by running the audio tests. Confirm that audio is routed correctly to each speaker. If not, remap the pin layout again using hdajackretask and retest again.

Screen shots of success:

As attached:

Success, at last!

Now enjoy great surround sound from your sound card.

]]> <![CDATA[PiSound, The Audio Card For The Raspberry Pi]]> https://pauljacobevans.wordpress.com/2017/03/20/pisound-the-audio-card-for-the-raspberry-pi/ Tue, 21 Mar 2017 02:34:16 +0000 Paul Jacob Evans https://pauljacobevans.wordpress.com/2017/03/20/pisound-the-audio-card-for-the-raspberry-pi/ Kids today are being loud with their ‘drum machines’ and ‘EDM’. Throw some Raspberry Pis at them, and there’s a need for a low-latency sound card with MIDI and all the other accouterments of the modern, Skrillex-haired rocker. That’s where PiSound comes in.

Of course, the Pi already comes with audio out, but that’s not enough if you want to do some real audio processing. You need audio in as well, and while you’re messing around with that, adding some high-quality opamps, ADCs, DACs, and some MIDI would be a good idea. This is what the PiSound is all about. …read more http://pje.fyi/NgtZJv

Paul Jacob Evans

]]> <![CDATA[Linux Active Crossover is working!]]> https://therationalaudiophile.wordpress.com/2015/07/11/linux-active-crossover-is-working/ Sat, 11 Jul 2015 15:46:50 +0000 therationalaudiophile https://therationalaudiophile.wordpress.com/2015/07/11/linux-active-crossover-is-working/ [Update: now running on a fanless Bay Trail processor]

After a few evenings of half-hearted attempts to port my Windows code and make the changes needed to run on Linux, I finally got my head around what was needed, and it works! Unfortunately I’m not at the house where the amp and speakers are so I can’t try it ‘in anger’ but at least I can tell that I’m getting what sounds like correctly-filtered Spotify or CD from the three stereo outputs.

On a ten year old Dell GX520 it’s using about 16% of the CPU, and when you add in Spotify at about another 16% plus the snd-aloop driver and all the other stuff going on in an internet-connected PC, it comes to about 40% CPU, which is a bit higher than I had hoped – there’s a tiny amount of fan noise. Maybe there is scope to improve the efficiency of the crossover software: at the moment I am reading and writing 32 bit integers to/from the sound cards (one is a dummy sound card of course) but doing all the processing in floating point which therefore involves converting each sample twice with a potentially expensive operation. Maybe this can be speeded up. And I can always find a faster, cooler PC of course.

[13/07/15] In response to a comment, the point of all this is not just to implement basic crossover filtering, but to correct the drivers’ individual responses based on measurements, producing zero phase shift for each driver, and therefore perfect (or as close as possible) acoustic crossovers and zero overall phase shift. EQ such as baffle step correction is overlaid onto the filters’ responses without costing anything extra in CPU power. Individual driver delays are also added. I am not claiming this is unique, but nor is it commonplace. In terms of an active crossover it is the no-compromises version.

I have had this system working for a couple of years on a Windows PC, but Linux will be a cheaper and more elegant solution.

[UPDATE 18/0715] I have it running with the speakers with a choice of two sound cards: Asus Xonar DS and Creative X-Fi. It’s just a case of changing a few characters in the xover config file.

The control loop algorithm for maintaining the average sample rate at input and output (and avoiding any resampling) is an interesting problem to solve and I have had fun trying different algorithms based on PID loops and plotting the result out as a graph. The output sample rate is fixed, set by the card, and has to be inferred from the time between calls to send chunks of data to the output card but there will be a level of jitter on this due to the other things that the multi-threaded program is doing. We know the precise sample rate at the input (the snd-aloop loopback driver) because we are setting it. The aim is to keep the difference between number of samples read and number of samples output to the DACs at a constant level, but as we are sending and receiving chunks of data the instantaneous figure is fluctuating all the time. I presume that similar calculations are being performed in the adaptive resampling that would be usual when connecting together digital audio systems with differing sample rates – the difference being that this would affect the audio (subtly, but it undeniably would), while the aim of my scheme is that the timing adjustments merely affect the fill level of a FIFO, the sample rate being rigidly fixed and defined by the DAC.

[UPDATE 31/07/15]

Feeling confident, I bought an Asus Xonar U7 USB 7.1 sound card. This is based on the CM6632A chipset. I got it working but… trying to set the format to signed 32 bit within my program failed when addressing the device as “hw”. It also failed with S24_3LE and various other sample formats. However, 16 bit was accepted. Consulting the web, people commonly seem to have this issue with both CM6631A and CM6632A on Linux, and their workaround is simply to use “plughw” instead. However, if the “hw” device rejects a format, then, supposedly, the hardware cannot support it. All the “plughw” device does is automatically allow the OS to convert samples from the format you are using into one that the card can use. So I have a feeling that the card is only running in 16 bit mode, regardless of what my code is sending it.

If an application chooses a PCM parameter (sampling rate, channel count or sample format) which the hardware does not support, the hw plugin returns an error. Therefore the next most important plugin is the plug plugin which performs channel duplication, sample value conversion and resampling when necessary.

http://www.volkerschatz.com/noise/alsa.html

[03/08/15 UPDATE] Got back to the house where my system lives after the weekend, and was able to try my Asus Xonar U7 again. This time it accepted S24_3LE! Could this be the issue with hot-plugging versus not hot-plugging that other people on the web have seen? I have a feeling that my previous tests were with the U7 hot-plugged into a PC that was already on. Anyway, I now seem to be in business with the U7 and it sounds good.

]]> <![CDATA[Trying Linux]]> https://therationalaudiophile.wordpress.com/2015/03/09/trying-linux/ Mon, 09 Mar 2015 09:44:07 +0000 therationalaudiophile https://therationalaudiophile.wordpress.com/2015/03/09/trying-linux/ UPDATED 16/03/15ubuntu-logo-8647_640 Approximately every two years I find myself inspired to have a go with Linux. I install Ubuntu on an old PC and congratulate myself on having finally made the right choice. Everything works fine: all the devices are auto-detected correctly, and although the graphics and text are a bit lumpy, it looks as though it can do everything Windows can do. It never lasts. Within a short time I try to do something beyond the basic web surfing and word processing and it doesn’t quite work. So I go to the web, and of course there’s usually a solution buried in a forum somewhere, and it invariably involves editing a config file. But along the way I may have found several other ‘solutions’ that didn’t work, and for each I maybe edited a different file or changed something using some little app I’ve installed. At the end, even though the system may be working, I am never quite sure how I got there, nor confident I could reproduce the same working system on another PC.

Well, the time has come again, and I am typing this using the latest version of Ubuntu. Everything is wonderful so far, and even Spotify is running flawlessly. Specifically, though, I want to get my active crossover system working on Linux, not Windows. My experience with Windows 7 running on slightly older PCs is not good. I have a laptop approximately 5 years old which will grind almost to a halt for several minutes every day, performing some sort of scan of itself, and I don’t know enough to do anything about it. The desktop PC that I use for the active crossover is slightly better, but it, too, takes quite a while to ‘warm up’ and is also prone to the occasional glitch while playing music, due to deciding to update its anti-virus database – I am sure it was not a problem with Windows XP. In contrast, running Ubuntu on an older desktop PC without much RAM, the experience is one of ‘solidity’. I am not experiencing the operating system going AWOL for several seconds at a time. But it comes at a price. I really, really don’t want to have to understand the details of any operating system, and Windows is good for the person who maybe wants to dip into a bit of programming (a distinctly different activity from IT) without having to worry too much about the really low level details. Windows feels as though it is ‘self-healing’. Every time the PC is turned on it starts scanning itself, checking for inconsistencies, downloading updates. New hardware is detected automatically and the user never edits configuration files. Ubuntu feels a little different. By all means correct me if I am wrong, but the impression I get is of a system that is dependent on lots of configuration files that are not hidden from the user. Of course these files get changed by the operating system itself (just as Windows must change its hidden configuration files) and there are little applications that you can install that simplify changing the parameters of various sound cards, say (more on this later). But occasionally the configuration files must be edited by the user using a text editor. One typo, and the PC may refuse to boot!

As I mentioned, I am hoping to run my active crossover stuff on Linux, not Windows. In order to achieve this I must loop continuously doing the following:

  1. Extract a chunk of stereo audio from an ‘input port’ that receives data from my application of choice (media player, Spotify etc.)
  2. Assemble the data into fixed-size buffers to be FFT-ed.
  3. Process with FIR filters to produce a separate, filtered output for each driver.
  4. Inverse FFT.
  5. Squirt the results out to six or eight analogue channels, or if feeling ambitious, HDMI (that would be the dream!).

It’s a very specific, self-contained requirement. I can handle numbers 2 to 4, no problem. 1 and 5 are the tricky ones, and seem to be a lot trickier than they, perhaps, might be. They weren’t all that easy in Windows, either, but I eventually came up with a scheme that kind of worked.

Here’s where it gets very specific: under XP I was able to use a single Creative X-Fi surround sound card as both the ‘receptacle’ for PC audio which I could then access with my application, and also as the multichannel DAC that my application could squirt its output to. Under Windows 7 the driver for the sound card was ‘updated’ and I could no longer access it as the receiver for general PC audio – I could still have used it for S/PDIF, analogue Line In etc., however. In the ideal world, the ‘receptacle’ would just be some software slaved to the output sample rate, I think, but I don’t know how to create such a piece of software – it would appear to Windows to be a driver I would guess. I could buy a piece of software called Virtual Audio Cable but I could never be sure whether that would always be re-sampling the data, and I’d rather avoid that. In the end, I used a method that I knew would work: I slaved a ‘professional’ audio card to the X-Fi using S/PDIF from the X-Fi. The M Audio 2496 can slave its sample rate to the S/PDIF (using settings in the M Audio-supplied configuration application) so I was able to send PC audio to the M Audio and my application could extract data from its ‘mixer’ at the same sample rate. Keeping the input and output on separate cards like this has some advantages when it comes to making measurements of the system while it is working, I think.

As a start I will probably try to do the same thing under Linux. I am attempting to use an Asus Xonar as the multichannel DAC, and another M Audio card I had lying around as the slaved source. It’s almost certain that I could achieve the objective without a second sound card, but I really don’t know how to do it [update 30/06/15: maybe I do know how to do it now]. Linux audio seems to have several ‘layers’ that I don’t understand (but as yet I have no view of them as layers, more as spaghetti). Really, I would like not to have to know anything about them at all, but this seems unrealistic. I have established the following:

  • I can do lowish-level audio stuff using the Alsa API. I can refer to specific cards by names that I can bring up with certain command line (shell) queries. Are these names guaranteed to stay the same in between boots? I don’t think so, but there are ways of editing the config files to associate names I choose to specific cards – I think.
  • There is a highly comprehensive system called JACK that allows “JACK-aware” programs to have their audio routed via a user-configurable patchbay. It can handle re-sampling between separate cards transparently. Brilliant, but I don’t think Spotify is “JACK-aware” for example so I’m not bothering with it. [Update 30/06/15: I want to avoid any form of re-sampling anyway]
  • Ubuntu has PulseAudio installed already (I think) and using an application (that I had to install) called Pavucontrol I can direct Spotify, and presumably other apps, to send their outputs to any of the sound cards in the system. Does this get written to a file and saved when I exit it? I think so. PulseAudio may be the thing I need, possibly being capable of creating software “sources” and “sinks”. But is it always resampling the audio to match sample rates even when that is not needed? More investigation needed. [Update 30/06/15: Pulseaudio cannot be guaranteed not to resample. I have removed it from the machine].
  • I installed a little program called Mudita24 that gives me most of the functionality of the app that is supplied for M Audio cards under Windows. It will let me slave the M Audio to S/PDIF. But without a lot of rummaging around on the web, finding this solution was not obvious. Will the results be saved to a file so I don’t have to call this up every time? I don’t know. [Update 30/06/15: the M Audio-compatible drivers don’t seem to work properly. I have abandoned this idea].
  • I found a “minimal” example program that can send a sine wave to an output via Alsa. The program is anything but minimal and allows the user to select from a large number of alternative sample rates, bit depths etc. etc. and has copious error reporting. My version of “minimal” is much shorter! I adapted the program for eight channels, and am sending a separate frequency to each of the Xonar’s outputs. It seems to be working quite solidly. I can’t be absolutely sure that the Xonar isn’t applying surround sound processing to the signals yet, though. Question: should I be programming using Alsa or PulseAudio? [Update 30/06/15: answer is most definitely ALSA only].

I don’t mind if everything is low level, nor do I mind if the operating system handles everything for me. What I am not keen on is a hybrid between the operating system doing some things automatically, and yet having to manually edit files (I haven’t done that yet, though) or having to install little apps myself. How are they all tied together? I don’t know.

UPDATE 10/03/15 Installed Ubuntu on my erratic Windows 7 laptop. On the hard drive I had to delete the ‘HP Tools’ partition to do it, as a PC can only have four partitions, apparently, and HP had used all four to install Windows – the things you learn, eh?

For the things I use the laptop for mainly, Ubuntu is knocking Windows 7 into a cocked hat. It actually responds instantly and doesn’t hang for tens of seconds with the disk light on constantly and the mouse pointer frozen. It’s taking some getting used to!

UPDATE 15/03/15 It is becoming clear to me that there is only one sensible solution for what I am trying to achieve (an active crossover / general DSP system under my control that can be applied to any source including streaming) that is guaranteed not to resample the data, nor is dependent on sound card-specific features, or needs two sound cards. Let me run this by you:

  • Media player apps need something that looks like a sound card to play into. Some apps will only play into whichever card is set as the default audio device.
  • If it’s a real sound card that’s being played into, I need to extract the data before it reaches the analogue outputs. This just may not be possible with many sound cards, and it is impossible to know without trying the card – no one cares about this issue normally.
  • I process the data into six or eight channels and then I need to squirt the results out to, effectively, some DACs (or HDMI). This is most likely a real, physical multi-channel sound card.
  • I believe that the media player’s sample rate is defined by the sound card it is playing into. If so, this is akin to asynchronous USB mode i.e. the media app is slaved to the sound card’s sample rate.
  • I would like to avoid sample rate conversion (and this would still be needed to convert between 44.09999 kHz and 44.10001 kHz i.e. there is no such thing as “the same sample rate” unless they are derived from the same crystal oscillator).

There is a Linux driver called snd-aloop which can act as a virtual audio node, recognisable by media player apps as a sink, but also recognisable by other apps as a recording source. I could send media player output into this virtual device, recognise it as a source for my application, process the data and send the multi-channel audio to a consumer-level DAC card without it needing any special features. However, there is a subtle problem: aloop’s sample rate is derived from the system-wide “jiffies” count. It will not match the sample rate of the DAC card even if they are both nominally 44.1 kHz.

I see just one sensible solution: I have to modify the aloop code so that, when the information is available, it gets its sample rate synchronisation from the DAC card. I could either modify aloop and send it this synchronisation information via a ‘pipe’ or shared memory (if that’s possible) from my active crossover application, or I can make my active crossover application a virtual sound card driver itself. Either way, I would need to register the driver with the system so that it can be set up as the default audio device (using the usual GUI-based sound preferences).

To any Linux programmers out there: does this sound sensible and do-able?

More later.

Update 30/06/15: It seems that there is an updated version of the snd-aloop driver which incorporates a dynamically-adjustable sample rate via the Alsa PCM interface. This could be precisely what I need.

]]> <![CDATA[Digger Dan, Sound Engineer?]]> https://diggerdanandthedirtbrigade.wordpress.com/2015/01/16/digger-dan-sound-engineer/ Fri, 16 Jan 2015 12:46:28 +0000 Digger Dan https://diggerdanandthedirtbrigade.wordpress.com/2015/01/16/digger-dan-sound-engineer/ Six months..yikes, that’s too long.

I’d like to say it is partially full time work that has kept music in the background. I think another reason is the extended bed rest in 2011-2012 destroying the momentum of playing almost every day. That has come back in flashes though.

I think it is something more that is keeping me from “pulling the trigger” and working on this project. No, not the money, although that is certainly a factor. I think it’s just a belief in myself and completely taking responsibility for the project from start to finish. There is a hesitancy with recording. I’m not 100 percent sure of what I am saying or ..admitting, but I think it is maybe something deeper.

Well, where do I go from here?

Keep assessing whether what I am saying is really true and try some new methods. I’ll save the assessing for another time on the virtual counselors couch and tell you about the plans in the works to get things going.

First, I need a computer at home that I can seamlessly work between the VAMS studio and home. At VAMS they use a MAC DAW (Digital Audio Workstation) and I used to have a 5,1 iMac at home, but I sold it to good ‘ol Joe Durgo. I want to write a song about him one day called No Ordinary Joe, but that is another matter. Another lotta gray matter actually!

So I sold the comp to Joe so we could work together on some video projects for him and I bought another Mac for my wife, a Macbook from around 2010. The idea was that we were going to share the Macbook between her video editing and my music recording but it is not feasible. She is very passionate about her DIYs on her channel (not to mentioned talented!), and I don’t want to start “booking time” around spontaneous creativity. So I need my own Mac.

I went through a spell of using Linux for music editing over the last few months but it was a fail. I love Linux, I mean love it, but for music recording, I just don’t have the capabilities. I have made friends with a very bright young fellow that could help do some IT for me on Ubuntu Studio or maybe even Studio 1337, but that would require having him ready to repair at a moments notice. Too much to ask a new friend.

So I’ve been combing craigslist, Free Geek, and a few other online sources for a Mac and had a good bite the other day on one. I just need to raise another 100 and I’m safe to get it. So that is one obstacle that will be solved soon.

The second obstacle is getting into “playing shape” and my plan is to bring the guitar into work and play for a half hour at lunch. I’m working eight hours a day but they are mixed between all kinds of times so I often find I’m working in the evening a bit, some on weekends and it is bleeding into my plan of getting into shape at night. That will eventually happen when I wrestle the work dog to the ground and pin him, but until then, it’s finding times that are a bit more unconventional. I think the acoustics at work will be good…it is a storage facility!

I’ve also decided that I need to do this myself. I guess that is heavily implied by saying I’m looking for a Mac at home, but it needs to be stated. The best I can do with having someone else help me is either pay 40 dollars per hour, which is a great deal for professional sound engineering, get a buddy to work for nothing (not acceptable), or to do it myself. I just am a bit daunted by the learning curve. I seem to be on many learning curves these days. So many I’d say I’m bending dizzy. But, what’s another one but a good opportunity to learn. Plus, there are always people who will help out a bit here and there.

So it’s part time at VAMS on my own (with the odd little bit of help from Dave or Graham possibly), and time at home on my own. That’s the plan!

Now, to the “not believing in myself” area..well, I wish I never wrote that and since I have a no edit policy going right now on this blog, I won’t take it out. But I wonder about that part…

]]> <![CDATA[Tocando musica desde dispositivos USB (huevonamente)]]> https://tumbaburrosopensurs.wordpress.com/2014/11/13/tocando-musica-desde-dispositivos-usb-huevonamente/ Thu, 13 Nov 2014 14:19:47 +0000 Web-0n https://tumbaburrosopensurs.wordpress.com/2014/11/13/tocando-musica-desde-dispositivos-usb-huevonamente/ Pues resulta que mi audio desde hace bastantes años lo maneja el Linux, por medio de un adaptador bluetooth envia el audio a un modular a travez de un dongle, tengo una tornamesa USB, con la cual tengo rato usandola para grabar en Audacity o en Cooledit via wine sin problemas, pero queria escuchar algunos viniles antes de grabarlos y me di un buen tiro queriendo hacer un bypass desde el tocadisco al estereo… uno de los problemas fue que mi distro es ya bastante antigua, asi que todos los repos ya no estan disponibles y la procrastinacion de respaldar e instalar una distro fresca me obligo a hacerlo con las herramientas sencillitas que estan en la instalacion default, porque por ejemplo no me permitia instalar el Pulse audio manager (paman) para ver los devices graficamente…

En muchos foros mencionaban que esto se hace facilmente con el probrama jack… pero a mi no me funciono…

Lo que hice fue lo sig.

Verificar que se detecten los dispositivos…

cash@becca ~ $  dmesg | tail
[   35.766861] Bluetooth: HIDP (Human Interface Emulation) ver 1.2
[  488.458009] input: 00:10:61:00:D9:2E as /devices/virtual/input/input10
[ 2360.716035] usb 2-1: >new full-speed USB device number 2 using uhci_hcd
[ 2361.041042] usb 2-1: >New USB device found, idVendor=08bb, idProduct=2900
[ 2361.041048] usb 2-1: >New USB device strings: Mfr=1, Product=2, SerialNumber=0
[ 2361.041053] usb 2-1: >Product: USB Audio CODEC
[ 2361.041058] usb 2-1: >Manufacturer: Burr-Brown from TI
[ 2361.048168] input: Burr-Brown from TI               USB Audio CODEC  as /devices/pci0000:00/0000:00:1d.0/usb2/2-1/2-1:1.3/input/input11
[ 2361.048329] hid-generic 0003:08BB:2900.0004: >input,hidraw3: USB HID v1.00 Device [Burr-Brown from TI               USB Audio CODEC ] on usb-0000:00:1d.0-1/input3
[ 2361.103419] usbcore: registered new interface driver snd-usb-audio

Obtener el nombre para pulse Audio del player

cash@becca ~ $  pactl list short | grep USB
4    module-alsa-card    device_id=”1″ name=”usb-Burr-Brown_from_TI_USB_Audio_CODEC-00-CODEC” card_name=”alsa_card.usb-Burr-Brown_from_TI_USB_Audio_CODEC-00-CODEC” namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes card_properties=”module-udev-detect.discovered=1″
0    alsa_output.usb-Burr-Brown_from_TI_USB_Audio_CODEC-00-CODEC.analog-stereo    module-alsa-card.c    s16le 2ch 44100Hz    SUSPENDED
0    alsa_output.usb-Burr-Brown_from_TI_USB_Audio_CODEC-00-CODEC.analog-stereo.monitor    module-alsa-card.c    s16le 2ch 44100Hz    SUSPENDED
1    alsa_input.usb-Burr-Brown_from_TI_USB_Audio_CODEC-00-CODEC.analog-stereo    module-alsa-card.c    s16le 2ch 44100Hz    RUNNING
0    alsa_card.usb-Burr-Brown_from_TI_USB_Audio_CODEC-00-CODEC    module-alsa-card.c

Reproducir a travez de un pipe

pacat -r -d alsa_input.usb-Burr-Brown_from_TI_USB_Audio_CODEC-00-CODEC.analog-stereo | pacat -p

Y voila! hagase la musica… es importante destacar que hay un delay en entrar como de 4 segundos, pero los acetatos que acabo de comprar estan sonando…

]]> <![CDATA[The perfect transport pt. 3 - software]]> https://stommager.wordpress.com/2013/08/23/the-perfect-transport-pt-3-software/ Fri, 23 Aug 2013 17:31:37 +0000 stommager https://stommager.wordpress.com/2013/08/23/the-perfect-transport-pt-3-software/ Dell Mini 9Laptops, or any personal computers for that matter, are not designed to be used as a high-end audio transports. It just so happens that they beat probably every dedicated audio transport in terms of price. Therefore it’s worth putting some effort in proper configuration of the former to get the quality of the later for a fraction of the price. That’s exactly what I did when my old CD player failed. Instead of buying a new one, I’ve invested in used netbook. This post summarizes my experiences in the software setup.

When I decided on buying Dell Mini 9, I wasn’t thinking much about the software. I just thought that I would install the bundled Windows XP, foobar2000 and listen happily ever after. But when I finally got the netbook I started to wonder… is Windows is the best OS for my music player? Of course there are number of articles on the Internet describing how to get bit-perfect audio out of Windows into an USB DAC, but I just couldn’t shake off this feeling of scepticism. You know how it is, Windows is designed for the lazy users, for the ones that want everything to work automatically, without any input from their part. This inevitably results in very complicated structure and limited configuration capabilities. In particular, I was afraid that although I would install and configure the proper software, Windows would still use some internal resampler or other treacherous process and ruin my efforts. I don’t trust folks at Microsoft who always seem to think they know what’s best for their users.

Then I discovered that ALSA finally implemented support for my DAC: E-MU 0404 USB audio interface. I was even more pleasantly surprised to find out that my external sound card worked flawlessly right out of the box with Ubuntu 12.04 LiveCD (Windows 7 still requires additional driver setup in order to use the E-MU). I would never guess that Ububtu will have better hardware support than Windows… times change. Everything looked really promising until I got into details of audio configuration. It turned out to be highly complicated with many different modules communicating back and forth, doing all sorts of crazy conversions and mixes. I was really disappointed because that’s exactly what we must avoid in order to get bit-perfect stream into the DAC. The only comfort was that, unlike Windows, Ubuntu gives reliable manual control and that the knowledge of it’s inner workings is widely available online. Therefore I decided to stick with Ubuntu.

I quickly identified that the main source of troubles is a component called PulseAudio. It’s main purpose, as I understand it, is to take sound streams from all sources, convert them to a common format, mix them together and pass to a component responsible for hardware (soundcard) access. This is reasonable, since typical user uses different sound inputs, sometimes simultaneously. But, as already mentioned, I’m no typical user and I don’t intend to use my Dell Mini as a typical computer. In my case, there will be only one source of sound and so no server/mixed functionalities are needed.

The question is: is PulseAudio really affecting the sound quality? Or maybe it is transparent to the audio stream if it’s coming from one source only? Well, I found some incriminating evidence in the system. First of all, PulseAudio has a configuration file with a settings that suggests, that it has only one output format. This doesn’t seem right. My audio interface supports many different sound formats, compatible with vast majority of audio files. When I play 24bit 48kHz file I expect that it will be transparently sent into by E-MU, which supports this format amongst many others. Unfortunately PulseAudio has this single fixed output format and it converts my file into the default 16bit 44,1kHz before sending it further. I’ve inspected E-MU’s running mode and found that indeed it is always working in the PA’s output format regardless of the played file format. So, the software conversion is a fact. Whether it inflicts the sound quality I cannot say as I haven’t done proper testing, but the point is it should not be there.

There are two main ways to remove PulseAudio from the processing chain. The first one is to uninstall it from the system or to use a simpler version of the system that just doesn’t include it by default. The second is to find the proper player capable of communicating directly with lower segments of audio processing chain, bypassing the default, evil mixer. I’ve decided to try out the later first. My favourite forum, head-fi.org has numerous threads on Linux audio players. I’ve tried some of the proposed names and found one to be particularly good, the DeaDBeeF.

It’s very simple, which is an asset for me. I’d say that it is written well. It doesn’t hang, it responds quickly, it plays gapeless tracks flawlessly, but most importantly it has the right configuration capabilities, for instance we can (and should!):

  • select the default output, in particular select ALSA instead of the default PulseAudio
  • set the output device
  • remove all DSP plugins
  • turn off ALSA’s resampling

The output devices come in many variants as ALSA recognizes many ways in which it can interact with the same hardware. In my case, the best option was “E-MU 0404 USB: Direct hardware device without any conversions”. This already sounds like music to my ears.

DeaDBeeF config

After setting everything up I repeated my test and found that this time E-MU’s running mode corresponds to the audio file format. Additional confirmation comes from the fact, that music’s volume is now totally independent of the system’s volume control and  PulseAudio control panel shows no playback when in fact the music plays. At this point I might very well uninstall the damn thing, but since it is not recommended by the Ubuntu team I decided not to.

At the moment of writing this post I’ve already made numbers of listens on this setup and I must say that I’m glad of all the decisions I’ve made along the way. I made a comparison: played the same CD through my CD player (fortunately it’s not entirely broken) and Dell Mini and switched the input on my E-MU. The result was that I had a feeling that the new system sounded a bit more detailed, but I’m not sure if I would tell them apart in a blind test. Theoretically the SQ should be a better (due to the fact that DAC is now synchronized with it’s internal clock and that we have eliminated the CD player’s potentially lossy error correction), but I’m not sure if I can hear the difference. Nevertheless I’m very glad that I have managed to find a high quality transport, a worthy replacement and that I stayed within my budget of 150$ (the actual unit cost was merely 85$, but I had to buy additional storage for the FLAC files for 47$). Functionally there are many advantages of this setup with only one small disadvantage: I have to rip every new CD. Other than that, all is convenient, reliable and works very fast. Real pleasure to listen to.

]]> <![CDATA[New Ardour Logo]]> https://thorwil.wordpress.com/2013/03/04/new-ardour-logo/ Mon, 04 Mar 2013 21:00:45 +0000 thorwil https://thorwil.wordpress.com/2013/03/04/new-ardour-logo/ Ardour is an application for recording, editing and mixing music. It is licensed under the terms of the GPL 2.

The upcoming 3.0 release seemed like a good opportunity to take another look at the logo I designed in 2006. A selection of drafts from back then, ending with the final design:
ardour_process_old

I had to ask myself: Is this logo (still) appropriate for Ardour?

The upcoming 3.0 release will be a digital audio and MIDI production application, available for Linux and Mac OS X. It is designed for frequent and prolonged use, being able to deal with huge amounts of material, complex signal pathways, precise and intense editing. Reliability, correctness and precision are of utmost importance.

The logo should take a matching stance, be sharp and have a strong presence. I think the old version does a fine job in this regard. It also happens to be well established and liked by the community (of course not by everyone). Back then I decided to use a free-form wave shape, less stylized, more realistic. Now I think a shape with even subdivisions will make the logo appear more precise.

I worked my way through variations of the curves that describe top and bottom of the wave, the number of teeth, their shape, relative height of the type and its consequences on letter spacing:

ardour_process_new
PDF of above image, in case you’d like to take a closer look.

ardour_logo_old_and_new

Application icons, first column are the old ones. I reduced the number of teeth for the smaller versions, keeping them at least 1 pixel wide.
ardour_app-icons

The new logo is already in use on the new website that went online about a week ago. I helped a bit with color selection, made a few suggestion and provided 3 icons:
record-edit-mix

]]> <![CDATA[Kick off Fedora 18 with a Bang!]]> https://crantila.wordpress.com/2013/01/15/kick-off-fedora-18-with-a-bang/ Tue, 15 Jan 2013 19:40:46 +0000 Christopher https://crantila.wordpress.com/2013/01/15/kick-off-fedora-18-with-a-bang/ I published the Musicians’ Guide for Fedora 18 earlier today (link).

Now that the Audio Creation SIG has our own spin (whether or not it’s official yet), I’d like to try moving forward with revisions to improve the Musicians’ Guide. I have virtually no spare time, and that’s my reality for the forseeable future, so instead I’d like to encourage everybody who uses Fedora’s music/audio software to contribute!

When you contribute to documentation for free software, you’re making it easier for somebody else to take their first steps into unknown territory. Comprehensive, accessible documentation is, in my opinion, one of the most important tools we have when trying to spread free software to new users.

Here are two easy ways to join in…

1.) Check out our list of bugs (here).

If you see something that you can do (or do part of), just add your proposed changes as a comment on the Bugzilla issue. Maybe you can rewrite an entire chapter or section, one paragraph a day. Later on, I’ll pick up your revised version, add the Docs Project-specific markup, and publish your changes with Fedora 19!

2.) Start using the Musicians’ Guide, and report issues (here).

The biggest challenge I have in maintaining this 270-page document is that, because I know there are *tons* of different areas for improvement, it’s difficult to know where to start. Feedback from real users is invaluable in helping to know where to spend my time. Even if it’s as simple as fixing a typo, clarifying a sentence, or including/excluding additional information, your feedback is extremely important.

And heck, let us know when things are going well, too!

]]> <![CDATA[Linux Audio Troubleshooting]]> https://itsecworks.com/2012/11/01/linux-audio-troubleshooting/ Thu, 01 Nov 2012 16:33:44 +0000 itsecworks https://itsecworks.com/2012/11/01/linux-audio-troubleshooting/ Imagine I had Linux Mint 12 and I connected my headphone to my linux machine, after disconnecting it I have lost my sound!?! I have alarm sounds in my monitoring system for many  companies and it was just silence, no alarms…
I had to check the settings with the alsamixer gui (alsamixer command) and something was muted automatically..
I have read many forums and wiki sites before I have found the easy solution, unmute everything. I made a script to collect my actual working audio settings for troubleshooting purposes. If it later goes wrong again, I run the script once more and I will compare the 2 outputs. If there is a difference, I will know at least what has changed, in which way should I go forward.

Do you have audio problems? Here you can start your troubleshooting:
https://help.ubuntu.com/community/SoundTroubleshooting
https://wiki.ubuntu.com/DebuggingSoundProblems

The commands used in my script:

output of the /etc/modprobe.d/alsa-base.conf file. For more info see http://alsa.opensrc.org/MultipleCards
output of the inxi -SAxc 0 command. inxi is a full featured system information script. It gives information about the audio drivers, cards. More info on http://code.google.com/p/inxi/
output of dpkg -l *pulse* and dpkg -l *alsa* commands to see what and which version is installed.
output of ps axfu | grep pulse and ps axfu | grep alsa command to see what runs actually.
output if aplay -l command. aplay is a command-line sound recorder and player for ALSA soundcard driver. It can list the soundcards and digital audio devices (here only the playback hardware devices).
output of lspci -v command. It lists all PCI devices inclusive the audio devices.
output of cat /proc/asound/version command. Shows the version of the audio driver.
output of head -n 1 /proc/asound/card0/codec* command.
output of amixer command. it is a command-line mixer for ALSA soundcard driver. With no arguments will display the current mixer settings for the default soundcard and device.
output of amixer controls command. It shows a complete list of card controls.

The script is here (rename it to sound_info.sh, it a bash script):

sound_info.sh

and my output is here:

audio_info.txt

]]> <![CDATA[Music: Gleshnor]]> https://thorwil.wordpress.com/2012/10/29/music-gleshnor/ Mon, 29 Oct 2012 10:16:09 +0000 thorwil https://thorwil.wordpress.com/2012/10/29/music-gleshnor/

Sugar-coated electronic rasp.

This project started as a test of Ardour 3’s new MIDI and synth-plugin features (still in beta). In this role, it served in uncovering and fixing a number of issues and grew into something a little more ambitious over time.

The kick, bass and lead all come from instances of Calf‘s Monosynth.

When thinking about what I could draw as a cover image, Driddee jumped into my mind. Similar to the music, creating this image was a test run, with Krita. Took a bit to get comfortable with it, but now I’m rather pleased. I need more practice, obviously :)

Other formats, the entire Ardour session as well as a track-by-track export (wavpack format) are available via archive.org.

Creative Commons License
Gleshnor and Driddee by Thorsten Wilms are licensed under a Creative Commons Attribution-ShareAlike 3.0 Unported License.


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]]> <![CDATA[Get on Board the Blues – Guicussion Remix]]> https://thorwil.wordpress.com/2012/07/17/get-on-board-the-blues-guicussion-remix/ Tue, 17 Jul 2012 08:13:05 +0000 thorwil https://thorwil.wordpress.com/2012/07/17/get-on-board-the-blues-guicussion-remix/ Dave Phillips recently published a great Blues track in a not so great mix and made the material available on request. Since others covered the gentle just bring out what’s there (by Fons Adriaensen) as well as tasteful addition of drums (by Jason Jones), I just had to do something a little different.

Online player, choice of different formats and the entire Ardour session packed up on archive.org.

All additions are created from the original material, no samples added. Both the original and remix have been produced with pre-release builds of what will become Ardour 3.0.

Lyrics

More music from Dave
More music from yours truly

]]> <![CDATA[m-audio transit on Ubuntu 12.04]]> https://linuxaudiolive.wordpress.com/2012/05/04/m-audio-transit-on-ubuntu-12-04/ Fri, 04 May 2012 17:51:52 +0000 lal https://linuxaudiolive.wordpress.com/2012/05/04/m-audio-transit-on-ubuntu-12-04/ I freshly installed Ubuntu 12.04 and the ubuntustudio meta package. To get the m-audio transit card working I followed my older post (m-audio transit and ubuntu linux), i.e. installing madfuload which comes in the repository and adjust the udev-rules. They do not work the way they come installed.

So I used the content from corrected-madfu-rules and saved it as /etc/udev/rules.d/41-madfuload.rules. The according file in /lib/udev/rules.d I left untouched. The “41” instead of “42” is there because it reads “…Pick a number higher than the rules you want to override, and yours will be used. …” in /etc/udev/rules.d/README.

Something hang on my machine when trying the outcome so I rebooted the machine, although this might not be necessary for everybody. After that the sound card works as it used to.

]]> <![CDATA[Second step F15 audio]]> https://zeboks.wordpress.com/2011/09/11/second-step-f15-audio/ Sun, 11 Sep 2011 16:11:05 +0000 myship https://zeboks.wordpress.com/2011/09/11/second-step-f15-audio/ Reconsidering what I said in the first post, I should warn insofar as using CCRMA repos, even for just a kernel install, can be misleading, considering the state of the whole update process. See this post.

It seems that the latency results for a normal 2.6.40.4-5.fc15.i686 are little bit worse than those of my Debian Sid install with a pengutronix 2.6.33.7.2-rt30-1-686, but they could be developed in time. Most important, they are far better then those of the CCRMA kernel for F15 without the init=upstart tag.

So I just stopped hacking this way for now, peace.

]]> <![CDATA[Fedora 15 install with sound]]> https://zeboks.wordpress.com/2011/09/07/fedora-15-install/ Wed, 07 Sep 2011 16:41:45 +0000 myship https://zeboks.wordpress.com/2011/09/07/fedora-15-install/ I am a Debian user, with a not so-short experience of Linux systems. Not expecially deep into the core of their secrets, as many others, I still do like to experiment with the huge possibilities of the different versions. Being a musician, I have finally afforded to install Fedora 15 on a 500 Gb USB disk.

I am now listening to some Sun Ra’s “Calls for All Demons”, after having set up the daemons of the system and, yes, it’s a quality audio. But before to get here I had to pay my dues, so I decided to give my contribution for others who may be interested to begin with Fedora as a system with a slant for music.

———-
First of all, if you install side-to an existing Linux-grub2 booting machine, do this (Link) and you’ll avoid to have to boot manually at the grub prompt, or trying fancy custom_40 ‘voicings’.

You may encounter pulse-audio problems. I did, and I decided to use the axe: I uninstalled it (pulseaudio and pulseaudio-alsa-plugins). It’s up to you and the card you have. I did it at the end after reading this alsa wiki, assuring me that the actual alsa system does not need config files to work at the basic level. True.
Subsequently, I think I’ll prefer to play a little with alsa better than  with pulse. Some time before I was asked from Kde system settings in Debian Sid to get rid of my alsa devices and stick on pulse and answered no, suspecting the proposal.

Finally, read carefully the Fedora Musician’s guide and be careful with CCRMA repos. For me the path has been the following: to have the kernel working but not their core package, to start. If you need low latency the normal Fedora kernel is not bad at all, yet, with planet CCRMA PAE kernel and the music software of the normal repositories, your deal can be made. Install your music software after you have the real-time kernel booted succesfully, maybe without the “core” CCRMA package, if you do not still know too well what you are doing.

If you do not have pulse audio, config-files of each application will permit you to point to alsa instead than pulse, that is becoming more and more the default user-end.

If your machine it’s not a dedicated machine and you need all your stuff, internet connection, etc., you will want to skip the init=upstart tag in your grub.cfg and tweak the rest with the help of this page, rosegarden is a good application to test the state of your system-timer, maybe you’ll have to use force=hpet in your grub.cfg.

It took me three days to tame the new personality at home, and it was not easy.  Now Debians and Fedoras are friends and Sun Ra orchestra is on “There will never be another you”, cool.

]]> <![CDATA[a "new to me" laptop for audio]]> https://ghostofjohntoad.wordpress.com/2010/07/12/a-new-to-me-laptop-for-audio/ Mon, 12 Jul 2010 12:52:18 +0000 the ghost https://ghostofjohntoad.wordpress.com/2010/07/12/a-new-to-me-laptop-for-audio/ For a while now I have been wanting to get a decent laptop for music and audio production, but do to funds I have had to hold off.  While working on my desktop works fine I really work best when I can be portable so a laptop suits my flow.  So move forward…

I recently received an HP 8510w laptop in desperate need of some TLC.  Someone had apparently spilled some coffee on it and it was dead when I got it.  I looked it over and found it also needed a hard drive, keyboard, and power supply.  I was skeptical if it was worth it.  Thanks to ebay and NewEgg however I was able to get it back up and running for about $200.  Not bad considering I couldn’t get anything quite nearly as good for double the amount.  It was a pain in the ass however dismantling everything and putting it back together, much harder than a desktop. Along the way I had issues with the thermal compound as well as the hard drive caddy that further complicated the journey.  Well onward,  Here are the quick specs:

  • Intel Core 2 Duo T7700 2.4 GHz
  • 4GB DDR2 RAM, up to 8GB possible
  • 250GB, 7200 RPM HD
  • NVIDIA Quadro FX 570M
  • Misc: Firewire, SD Card Reader, Type I/II PC Card slot, HDMI port

Seeing how this will be used mainly for audio/music applications this should be more than sufficient.  I will be using my Tascam us122 and Zoom H4n for getting audio into the computer.

I will be installing Linux, particularly Ubuntu 10.04, and then installing the Ubuntu Studio packages over top.

Linux is the choice for me for a variety of reasons, latency, cost, and the challenge for me personally to do things a little differently.

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