My thoughts on Slackware, life and everything

Category: Hardware (Page 1 of 4)

Sonic Pi 4.1.0 packaged for Slackware – and for live coding musicians

Sam Aaron worked long and hard towards a new major release of his Sonic Pi software, and kept us informed about his coding journey on Twitter. Eventually a 4.0.0 version saw the light of day, 17 months after the  final 3.x  version (3.3.1) came out. Sam writes really informative release notes, I encourage you to check them out on his release page. Lots of enhancements and fixes to be enjoyed.

You don’t know what I am talking about?

You’ve probably missed my last year’s blog post then.
Sam Aaron is a musician and a programmer, holding a PhD in Computer Science. He develops a really cool program called Sonic Pi, meant to educate people in making music and learning how to code at the same time.
Sonic Pi uses the engine of Supercollider, which is another live coding platform. But Sonic Pi makes the art of live coding – performing on stage or just having a blast creating music on your computer – very low-threshold. Out of the box, when you start the program, its GUI will show complete and working coding examples, a tutorial, and an intuitive interface to start producing music straight away:

I have created packages for sonic-pi 4.1.0, the latest release, and I guess I am the first to create 32bit packages for a Linux platform, since I had to teach the “vcpkg” program that 32bit Linux is actually a thing. You could have guessed it… vcpkg is a cross-platform library manager created by Microsoft (!!!) which Sonic Pi uses to compile the libraries it needs.
I’ll see if I can create a patch for 32bit Linux support and send it to them.

Anyway, a month ago Sam Aaron gave a lecture at Lambda Days 2022 in Poland, about Sonic Pi, and here is a link to a recording of that lecture. You’ll learn more by watching that video, than by reading my blog posts about it.

The new packages (for Slackware 15.0 and -current) can be downloaded from all the usual places, for instance my primary site, or the fast NL, UK or US mirrors.

I hope you are going to give Sonic Pi a try and that I’ll hear from you in the comments section below if you came up with a cool piece of audio.

And because Sonic Pi is contained in my Slackware Live DAW edition, I have also generated a new ISO for the Digital Audio Workstation. Get the “slackware64-live-daw-current.iso” ISO file, and copy it to a Ventoy USB stick or use the “” script to create a persistent USB stick.
This live environment should allow you to run Sonic Pi out of the box even if you have no idea how to configure the JACK audio server.

Have fun! Eric

I now have a US mirror for Slackware Live and other goodies

Thanks to an anonymous sponsor, I am now operating a physical server in a US data center with a 1 Gbps connection to the Internet.

This server addresses a complaint of many people who are trying to download ISOs of the Slackware Live Edition. My aka server is hosted in a Dutch datacenter in Amsterdam, and it looks like people outside Europe, in particular downloaders in Southern Pacific region, are experiencing terribly slow speeds when fetching content from that server.

My new US server is available at two main URLs:

  • is the go-to location for all content related to Slackware Live Edition.
  • (don’t be confused by the “.nl” domain… I do not own a “.us” domain unfortunately) is the resurrection of my old “taper” VM which did not survive the original release of liveslak… that taper buckled under the high demand caused by massive download traffic and I decommisioned it in favor of my French datacenter server “bear” which again was replaced with “martin” in Amsterdam.
    The new taper has mirrors for liveslak (exact same content as and also all Slackware release trees and ISOs, the ‘cumulative’ package repository, Mate SlackBuild (msb) and Cinnamon SlackBuild (csb), as well as my own package and multilib repositories.

In addition to the http access, these servers are also accessible via rsync: rsync:// and rsync://

I hope this will give you folks out there a good alternative mirror location. Let me know how you experience the download speeds.

Cheers, Eric

Configuring Slackware for use as a DAW

Welcome, music-making (Slackware) Linux-using friends. This article will be an interactive work and probably never finished. I am looking for the best possible setup of my Slackware system for making music without having to revert to a custom Linux Distro like the revered StudioWare.
Please supplement my writings with your expert opinions and point out the holes and/or suggest improvements/alternatives to my setup. Good suggestions in the comments section will be incorporated into the main article.
Your contributions and knowledge are welcomed!

What is a DAW?

In simple terms, a Digital Audio Workstation is a device where you create and manipulate digital audio.

Before the era of personal computing, a DAW would be a complex piece of (expensive) hardware which was only within reach of music studios or artists of name and fame. A good example of an early DAW is the Fairlight CMI (Computer Musical Instrument), released in the late 70’s of the previous century. This Fairlight was also one of the first to offer a digital sampler. The picture at the left of this page is its monitor with a light-pen input.

These days, the name “DAW” is often used for the actual software used to produce music, like the free Ardour, LMMS, or the commercial Ableton Live, FL Studio, Cubase, Pro Tools, etcetera.
But “Digital Audio Workstation” also applies to the computer on which this software is running and whose software and hardware is tailored to the task of creating music. To make it easier for musicians who use Linux, you can find a number of custom distributions with a focus on making electronic music, such as StudioWare (Slackware based), AV Linux (Debian based), QStudio64 (Mint based), so that you do not have to spend a lot of time configuring your Operating System and toolkits.

A ready-to-use distro targeting the musician’s workflow is nice, but I am not the average user and want to know what’s going on under the hood. What’s new… And therefore, in the context of this article I am going to dive into the use of a regular Slackware Linux computer as a DAW.

I wrote an earlier article where I focused on the free programs that are available on a Linux platform and for which I created Slackware packages: Explorations into the world of electronic music production.
That article provided a lot of good info (and reader feedback!) about creating music. What lacked, was a good description of how to setup your computer as a capable platform to use all that software on. Slackware Linux out of the box is not setup as a real-time environment where you have low-latency connectivity between your musical hardware and the software. But these capabilities are essential if you want to produce electronic music by capturing the analog signals of physical instruments and converting them to digital audio, or if you want to add layers of CPU intensive post-processing to the soundscape in real-time.
Achieving that will be the topic of the article.

Typical hardware setup

The software you use to create, produce and mix your music… it runs on a computer of course. That computer will benefit from ample RAM and a fast SSD drive. It will have several peripherals connected to it that allow you to unleash your musical creativity:

  • a high-quality sound card or external audio interface with inputs for your physical music instruments and mics, sporting a good Analog to Digital Converter (DAC) and adding the least amount of latency.
  • an input device like a MIDI piano keyboard or controller, or simply a mouse and typing keyboard.
  • a means to listen to the final composite audio signal, i.e. a monitor with a flat frequency response.

In my case, this means:

  • USB Audio Interface: FocusRite Scarlett 2i4 2nd Gen
  • Voice and instrument recording mics connected to the USB audio interface:
    • Rode NT1-A studio microphone with a pop-filter, or
    • Behringer C-2 2 studio condenser microphones (matched pair)
  • MIDI Input: Novation Launchkey 25 MK2 MIDI keyboard
  • Monitor: I use Devine PRO 5000 studio headphones instead of actual studio monitors since i am just sitting in the attic with other people in the house…
  • And a computer running Slackware Linux and all the tools installed which are listed in my previous article.

Configuring Slackware OS

We will focus on tweaking OS parameters in such a way that the OS essentially stays ‘out of your way’ when you are working with digital audio streams. You do not want audio glitches, drop-outs or experience out-of-sync inputs. If you are adding real-time audio synthesis and effects, you do not want to see delays. You do not want your computer to start swapping out to disk and bogging down the system. And so on.

Note that for real-time behavior of your applications, you do not have to install a real-time (rt) kernel!

The ‘audio’ group

The first thing we’ll do is decide that we will tie all the required capabilities to the ‘audio’ group of the OS. If your user is going to use the computer as a DAW, you need to add the user account to the ‘audio’ group:

# gpasswd -a your_user audio

All regular users stay out of that ‘audio’ group as a precaution. The ‘audio’ members will be able to claim resources from the OS that can lock or freeze the computer if misused. Your spouse and kids will still be able to fully use the computer – just not wreck havoc.

The kernel

A Slackware kernel has ‘CONFIG_HZ_1000’ set to allow for a responsive desktop, and ‘CONFIG_PREEMPT_VOLUNTARY’ as a sensible trade-off between having good overall throughput (efficiency) of your system and providing low latency to applications that need it.

So you are probably going to be fine with the stock ‘generic’ kernel in Slackware. One tweak though is going to improve on low-latency. One important part of the ‘rt-kernel’ patch set was added to the Linux source code, and that is related to interrupt handling. Threaded IRQs are meant to minimize the time your system spends with all interrupts disabled. Enabling this feature in the kernel is done on the boot command-line, by adding the word ‘threadirqs’ there.

  • If you use lilo, open “/etc/lilo.conf” in an editor and add “threadirqs” to the value of the “append=” keyword, and re-run ‘lilo’.
  • If you use elilo, open /boot/efi/EFI/Slackware/elilo.conf and add “threadirqs” to the value of the “append = ” keyword
  • If you are using Grub, then open “/etc/default/grub” in an editor and add “threadirqs” to the (probably empty) value for “GRUB_CMDLINE_LINUX_DEFAULT”. Then re-generate your Grub configuration.

CPU Frequency scaling

Your Slackware computer is configured by default to use the “ondemand” CPU frequency scaling governor. The kernel will reduce the CPU clock frequency (modern CPU’s support this) if the system is not under high load, to conserve energy. However, much of the load in a real-time audio system is on the DSP, not on the CPU, and the scaling governor will not catch that. It could result in buffer underruns (also called XRUNs). Therefore, it is advised to switch to the “performance” governor instead which will always keep your CPU cores at max clock frequency. To achieve this, open the file “/etc/default/cpufreq” in an editor and add this line (make sure that all other lines are commented out with a ‘#” at the beginnning!):


After reboot, this modification will be effective.

Note: if you want to use your laptop as a DAW, you may want to re-consider this modification. You do not want to have your CPU’s running at full clockspeed all of the time, it eats your battery life. Make use of this “performance” feature only when you need it.

Real-time scheduling

Your DAW and the software you use to create electronic music, must always be able to carry out their task irrespective of other tasks your OS or your Desktop Environment want to slip in. This means, your audio applications must be able to request (and get) real-time scheduling capabilities from the kernel.

How you do this depends on whether your Slackware uses PAM, or not.

If PAM is installed, RT Scheduling and the ability to claim Locked Memory is configured by creating a file in the “/etc/security/limits.d/” directory – let’s name the file “/etc/security/limits.d/rt_audio.conf” (the name is not relevant, as long as it ends on ‘.conf’) and add the following lines.

# Real-Time Priority allowed for user in the 'audio' group:
# Use 'unlimited' with care,a misbehaving application can
# lock up your system. Try reserving half your physical memory:
#@audio - memlock 2097152
@audio - memlock unlimited
@audio - rtprio 95

If PAM is not part of your system, we use a feature which is not so widely known to achieve almost the same thing: initscript (man 5 initscript). When the shell script “/etc/initscript” is present and executable, the ‘init’ process will use it to execute the commands from inittab. This script can be used to set things like ulimit and umask default values for every process.
So, let’s create that file “/etc/initscript”, add the following block of code to it:

# Set umask to safe level:
umask 022
# Disable core dumps:
ulimit -c 0
# Allow unlimited size to be locked into memory:
ulimit -l unlimited
# Address issue of jackd failing to start with realtime scheduling:
ulimit -r 95
# Execute the program.
eval exec "$4"

And make it executable:

# chmod +x /etc/initscript

The ulimit and umask values that have been configured in this script will now apply to every program started by every user of your computer. Not just members of the ‘audio’ group. You are warned. Watch your kids.

Use sysctl tweaks to favor real-time behavior

The ‘sysctl’ program is used to modify kernel parameters at runtime. We need it in a moment, but first some preparations.
A DAW which relies on ALSA MIDI, benefits from access to the high precision event timer (HPET). We will allow the members of the ‘audio’ group to access the HPET and real-time clock (RTC) which by default are only accessible to root.
We achieve this by creating a new UDEV rule file “/etc/udev/rules.d/40-timer-permissions.rules”. Add the following lines to the file and then reboot to make your changes effective:

KERNEL=="rtc0", GROUP="audio"
KERNEL=="hpet", GROUP="audio"

With access to the HPET arranged, we can create a sysctl configuration file which the kernel will use on booting up. There are a number of audio related ‘sysctl‘ settings that allow for better real-time performance, and we add these settings to a new file “/etc/sysctl.d/daw.conf”. Add the following lines to it:

dev.hpet.max-user-freq = 3072
fs.inotify.max_user_watches = 524288
vm.swappiness = 10

The first line allows the user to access the timers at a higher frequency than the default ’64’. Note that the max possible value is 8192 but a sensible minimum for achieving lower latency is 1024. The second line is suggested by the ‘realtimeconfigquickscan‘ tool, and increases the maximum number of files that the kernel can track on behalf of the programs you are using (no proof that  this actually improves real-time behavior). And the third line will prevent your system from starting to swap too early (its default value is ’60’) which is a likely cause for XRUNs.

Disable scheduled tasks of the OS and your DE

The suggestions above are mostly kernel related, but your own OS and more specifically, the Desktop Environment you are using can get in the way of real-time behavior you want for your audio applications.

General advice:

  • Disable your desktop’s Compositor (KWin in KDE Desktop, Compiz on XFCE and Gnome Desktop). Compositing requires a good GPU supporting OpenGL hardware acceleration but still this will put a load on your CPU and in particular on the application windows.


  • Plasma 5 Desktop
    • The most obvious candidate to mess with your music making process is Baloo, the file indexer. The first time Baloo is started on a new installation, it will seriously bog down your computer and eat most of its CPU cycles while it works its way through your files.
      If you really want to keep using Baloo (it allows for a comfortable file search in Plasma5) you could at least disable file content indexing and merely let it index the filenames (similar to console-based ‘locate’).
      Go to “System Settings > Workspace > Search > File search” and un-check “also index file content”
      If you want to completely disable Baloo so that you can not re-enable it anymore with any command or tool, do a manual edit of ${HOME}/.config/baloofilerc and make sure that the “[Basic Settings]” section contains the following line:
    • In KDE, the Akonadi framework is responsible for providing applications with a centralized database to store, index and retrieve the user’s personal information including the user’s emails, contacts, calendars, events, journals, alarms, notes, etc. The alerts this framework generates can interfere with real-time audio recording, so you can disable Akonadi if you want by running “akonadictl stop” in a terminal, under your own user account. Then make sure that your desktop does not auto-start applications which use Akonadi, Also, open the configuration of your desktop Clock and uncheck “show events” to prevent a call into Akonadi.
    • Compositing is another possible resource hog, especially when your computer is in the middle-or lower range. You can easily toggle (disable/re-enable) the desktop compositor by pressing the “Shift-Alt-F12” key combo.
      Note that if you use Latte-dock as your application starter, this will not like the absence of a Compositor. You’ll have to switch back to KDE Plasma’s standard menus to start your applications.


Selecting your audio interface

You may not always have your high-quality USB audio interface connected to your computer. When the computer boots, ALSA will decide for itself which device will be your default audio device and usually it will be your internal on-board sound card.

You can inspect your computer’s audio devices that ALSA knows about. For instance, my computer has onboard audio, then the HDMI connector on the Nvidia GPU provides audio-out; I have a FocusRite Scarlett 2i4, an old Philips Web Cam with an onboard microphone and I have loaded the audio loopback module. That makes 5 audio devices, numbered by the kernel from 0 to 4 with the lowest number being the default card:

$ cat /proc/asound/cards 
0 [NVidia_1    ]: HDA-Intel - HDA NVidia
                  HDA NVidia at 0xdeef4000 irq 22
1 [NVidia      ]: HDA-Intel - HDA NVidia
                  HDA NVidia at 0xdef7c000 irq 19
2 [USB         ]: USB-Audio - Scarlett 2i4 USB
                  Focusrite Scarlett 2i4 USB at usb-0000:00:02.1-2, high speed
3 [U0x4710x311 ]: USB-Audio - USB Device 0x471:0x311
                  USB Device 0x471:0x311 at usb-0000:00:02.0-3, full speed
4 [Loopback ]: Loopback - Loopback
               Loopback 1

Note that these cards can be identified when using ALSA commands and configurations, both by their hardware index (hw0, hw2) and by their ‘friendly name‘ (‘Nvidia_1’, ‘USB’).

Once you know which cards are present, you can inspect which kernel modules are loaded for these cards – The same indices as shown in the previous ‘cat /proc/asound/cards‘ command are also listed in the output of the next command:

$ cat /proc/asound/modules
0 snd_hda_intel
1 snd_hda_intel
2 snd_usb_audio
3 snd_usb_audio
4 snd_aloop

You’ll notice that some cards use the same kernel module. If you want to deterministically number your sound devices instead allowing the kernel to probe and enumerate your hardware, you’ll have to perform some wizardry in the /etc/modprobe.d/ directory.

Using pulseaudio you can change the default audio output device on the fly.
First you determine the naming of devices in pulseaudio (it’s quite different from what you saw in ALSA) with the following command which lists the available outputs (called sinks by pulseaudio):

$ pactl list short sinks
0 alsa_output.usb-Focusrite_Scarlett_2i4_USB-00.analog-surround-40 module-alsa-card.c s32le 4ch 48000Hz SUSPENDED
1 alsa_output.pci-0000_00_05.0.analog-stereo module-alsa-card.c s32le 2ch 48000Hz SUSPENDED
3 alsa_output.platform-snd_aloop.0.analog-stereo module-alsa-card.c s32le 2ch 48000Hz SUSPENDED
4 jack_out module-jack-sink.c float32le 2ch 48000Hz RUNNING
11 alsa_output.pci-0000_02_00.1.hdmi-stereo-extra1 module-alsa-card.c s32le 2ch 48000Hz SUSPENDED

You see which device is the default because that will be the only one that is in ‘RUNNING’ state and not ‘SUSPENDED’. Subsequently you can change the default output device, for instance to the FocusRite interface:

$ pactl set-default-sink alsa_output.usb-Focusrite_Scarlett_2i4_USB-00.analog-surround-40

You can use these commands to help you deciding which to use as your default device if you un-plug your USB audio interface, or if you want to let your system sounds be handled by an on-board card while you are working on a musical production using your high-quality USB audio interface – each device with their own set of speakers.

Connecting the dots: ALSA -> Pulseaudio -> Jack

The connecting element in all the software tools of your DAW is the Jack Audio Connection Kit, or Jack for short. Jack is a sound server – it provides the software infrastructure for audio applications to communicate with each other and with your audio hardware. All my DAW-related software packages have been compiled against the Jack libraries and thus can make use of the Jack infrastructure once Jack daemon is started.

Once Jack takes control, it talks directly to ALSA sound system.

Slackware uses the ALSA sound architecture since replacing the old OSS (open sound system) with it, many years ago. ALSA is the kernel-level interface to your audio hardware combined with a set of user-land libraries and binaries to allow your applications to use your audio hardware.
Pulseaudio is a software layer which was added to Slackware 14.2. Basically it is a sound server (similar to Jack) which interfaces between ALSA and your audio applications, providing mixing and re-sampling capabilities that expand on what ALSA already provides. It deals with dynamic adding and removing of audio hardware (like head-phones) and can transfer audio streams over the network to other Pulseaudio servers.
Musicians and audiophiles sometimes complain that Pulseaudio interferes with the quality of the audio. Mostly this is caused by the resampling that Pulseaudio may do when combining different audio streams but this can be avoided by configuring your system components to all use a single sample rate like 44,1 or 48 KHz. Also, in recent years the quality of the Pulseaudio software has improved quite a bit.

When Jack starts it will interface directly to ALSA, bypassing Pulseaudio entirely. What that means is that all your other applications that are not Jack-aware suddenly stop emitting sound because they still play via Pulseaudio. Luckily, we can fix that easily, and without using any custom scripting.

All it takes is the Jack module for Pulseaudio. The source code for this module is part of Pulseaudio but it is not compiled and installed in Slackware since Slackware does not contain Jack. So what I did is create a package which compiles just that Pulseaudio Jack module. You should install my “pulseaudio-jack” package from my repository. The module contains a library responsible for detecting when jack starts and then enables ‘source’ and ‘sink’ for Pulseaudio-aware applications to use.

The main pulseaudio configuration file “/etc/pulse/” already contains the necessary lines to support the pulseaudio-jack module:

### Automatically connect sink and source if JACK server is present
load-module module-jackdbus-detect channels=2

The only thing you need to do is ensuring that jackdbus is started. If you use qjackctl to launch Jack, you need to check the “Enable D-Bus interface” and “Enable JACK D-Bus interface” boxes in “Setup -> Misc”. See next section for more details on using QJackCtl.

In 2020 this is all it takes to route all output from your ALSA and Pulseaudio applications through Jack.

Easy configuration of Jack through QJackCtl

The Jack daemon can be started and configured all from the commandline and through scripts. But when your graphical DAW software runs inside a modern Desktop Environment like XFCE or KDE Plasma5, why not take advantage of graphical utilities to control the Jack sound server?

DAW-centric distros will typically ship with the Cadence Tools, which is a set of Qt5 based applications written for the KXStudio project (consisting of Cadence itself and also Catarina, Catia and Claudia) to manage your Jack audio configuration easily. Note that I have not created packages for Cadence Tools but if there’s enough demand I will certainly consider it, since this toolkit should work just fine in Slackware:

For a Qt5 based Desktop Environment like KDE Plasma5, a control application like QJackCtl will blend in just as well. While it’s more simplistic than Cadence, it does a real good job nevertheless. Its author offers several other very nice audio programs at like QSampler and the Vee One Suite of old-skool synths.

Like Cadence, QJackCtl offers a graphical user interface to connect your audio inputs and outputs, allowing you to create any setup you can imagine:

QJackCtl can be configured to run the Jack daemon on startup and enable Jack’s Dbus interface. Stuff like defining the samplerate, the audio device to use, the latency you allow, etcetera is also available. And if you tell the Desktop Session Manager to autostart qjackctl when you login, you will always have Jack ready and waiting for you.


Turning theory into practice

The reason for writing up this article was informational of course, since this kind of comprehensive detail is not readily available for Slackware. With all the directions shared above you should now be able to tune your computer to make it suited for some good music recording and production, and possibly live performances.

A secondary goal of the research into the article’s content was to gain a better understanding of how to put together my own Slackware based DAW Live OS. All of the above knowledge is being put into the liveslak scripts and Slackware Live Edition now has a new variant next to PLASMA5, SLACKWARE, XFCE, etc… it is “DAW“.
I am posting ISO images of this Slackware Live DAW Edition to and hope some of you find it an interesting enough concept that you want to try it out.

Note that you’ll get a ~ 2.5 GB ISO which boots into a barebones KDE Plasma5 Desktop with all my DAW tools present and Jack configured, up and running. User accounts are the same as with any Slackware Live Edition: users ‘live‘ and ‘root‘ with passwords respectively ‘live‘ and ‘root‘.

Why KDE Plasma5 as the Desktop of choice? Isn’t this way too heavy on resources to provide a low-latency workflow with real-time behaviour?
Well actually… the resource usage and responsiveness of KDE Plasma5 is on par or even better than the light-weight XFCE. Which is the reason why an established distro like Ubuntu Studio is migrating from XFCE to KDE Plasma5 for their next release (based on Ubuntu 20.10) and KXStudio targets the KDE Plasma5 Desktop as well.

You can burn the ISO to a DVD and then use it as a real ‘live’ OS which is fresh and pristine on every boot, or use the ‘’ script which is part of liveslak to copy the content of the ISO to a USB stick – which adds persistency, application state saving and additional storage capability. The USB option also allows you to set new defaults for such things as language, keyboard layout, timezone etc so that you do not have to select those everytime through the bootmenus.

If your computer has sufficient RAM (say, 8 GB or more), you should consider loading the whole Live OS into RAM (using the ‘toram’ boot parameter) and have a lightning-fast DAW as a result. My tests with a USB stick with USB-3 interface was that it takes 2 to 3 minutes to load the 2.5 GB into RAM, which compares to nothing if your DAW session will be running for hours.

Shout out

A big help was the information in the Linux Audio Wiki, particularly this page: In fact, I recommend that you absorb all of the information there.
On that page, you will also find a link to a Perl program “realtimeconfigquickscan” which can scan your system and report on the readiness of your computer for becoming a Digital Audio Workstation.

Good luck! Eric

Casting to your ChromeCast in Chromium

I have a Google ChromeCast, it is an extremely convenient gadget to stream all sorts of audio and video to your television, regardless of its make or brand.
Today I learnt something new, so let me tell you about that. But an introduction first.

I have a media collection at home, stored on a big disk in my server, and I have a Universal Media Server (UMS) running on my Slackware LAN server to get access to my audio and video files.

Usually from my Android Smartphone (I have always only bought HTC phones and currently it’s a HTC 10) I connect to my movies using BubbleUPnP, a cool application for which I actually bought a license. BubbleUPnP is a UPnP controller, meaning it does not necessarily play video by itself, but it is able to connect a media source (the UMS server) and a media renderer (like a television) and control the playback as a ‘man in the middle’.
For this to work, your television must understand UPnP and be able to announce itself as such on your network, and act as a media renderer. Many of the ‘smart’ televisions sold today have this capability, but many older televisions need a bit of help.
Enter the ChromeCast, a smart Linux-powered device with a wireless network card which connects to your television’s HDMI port. Once connected to your Wi-Fi network, it acts as a UPnP media renderer which means you can stream audio and video to it. However dumb, your television receives the HDMI input from the ChromeCast and will ‘just’ play back what it receives.
The BubbleUPnP app on my phone manages this with great success, and other apps will suddenly also have an extra button on-screen. Such as your phone’s Netflix and Youtube apps, which allows you to watch Netflix or Youtube on a big tv-screen instead of your puny little phone screen.

Now on a laptop or desktop computer, it is a bit of a different story. Your application must be able to act as a UPnP client to allow it to find a ChromeCast and stream media to it. On Linux, I could not find many applications that will cast media to a UPnP renderer. VLC is one good example. Chromium is another good example.

Or so I thought, but I was unable to get Chromium to find my ChromeCast… I tried from time to time, and when it would not work, I would simply take my phone and stream from there. But today, I thought ‘I need to fix this’. So, some searching around on the Internet and I found the fix.

All that’s needed is to enable “chrome://flags/#load-media-router-component-extension“… just copy and paste that bold text into your Chromium URL entry field and you get this:

The default setting called “Default” obviously is not sufficient, so I set the value to “Enabled” and voila! My Chromium browser finds the ChromeCast and I can watch fullscreen movies from a browser tab.

I hope someone sometime searches for and finds this piece of information and benefits from it.


Using your Slackware to control your Android

Linux and Android

No worries, even though this article is a side-step into the Android ecosystem, it is still on-topic for Slackware users.
I recently bought a Huawei Watch 2, when it was in a sale on Amazon. It’s the 4G version and long ago I had made a deal with myself that if a smartwatch with GPS and room for a SIM card would go below 300 euros, I was going to buy it. And so it happened, and the watch is a lot of fun. It’s incredibly light-weight and fits comfortably. Despite grumpy reviews it is a solid design and does not feel like cheap junk at all. I still need to find a provider who will allow me to use a second SIM card for my mobile subscription, so for the moment this watch is SIM-less. It talks to my phone over Wi-Fi and Bluetooth so nothing’s lost and I have full connectivity on the watch. The idea is to have a SIM in the watch so I can go outdoors for a long run without having to take my phone with me.

An Android Wear watch will pair with only one single phone. It’s quite picky about its friends. Therefore, if you switch phones and want to pair the watch to your new phone, it will wipe and reset itself to factory defaults. That was not what I wanted when I traded my four years old HTC One M8 for a HTC 10 phone. Yes, that too was a bargain sale – I got it for 272 euros which is roughly 65% of the regular price around here.

So I went searching for ways around my dilemma, and as it turns out: it is quite easy to de-couple an Android watch from its pal the phone, and let it pair with a new phone, while keeping its current settings and content. I am going to document that process here, because much of it is generic enough that you can use it to manage the internals of any Android device, like your phone.

The singularly important tool for interacting with Android devices is the Android Debug Bridge, or “adb” for short. It’s a program or rather a suite of tools that run on Linux, Windows and OS/X and allow you to execute commands remotely on the Android device. Commonly the bridge is used over a USB connection between Computer and Android device, but in this article I will show you that you can just as easily use a Wireless connection.

Installing Android Debug Bridge

First, let’s setup the Linux computer. In our case, that will of course be a Slackware computer, right? 🙂

ADB is a small part of the “Android SDK Platform-Tools” and in the past, this meant you had to install the complete SDK in order to gain access to adb and fastboot. Recently, Google has started making adb and friends available as a separate, much smaller (around 5 MB) ZIP download. The URL is static but the content will be refreshed from time to time.

  • Download the ZIP containing the platform-tools for Linux:
  • Extract the ZIP file to /usr/local (you need to be root):
    unzip -d /usr/local
    That leads to a directory /usr/local/platform-tools/ with a lot of stuff in it.
  • Create two wrapper scripts for the ‘adb’ and ‘fastboot’ binaries inside the extracted directory. This way, typing “adb” or “fastboot” will work in any directory:
    cat << EOT > /usr/local/bin/adb
    /usr/local/platform-tools/adb “$@”
    chmod 755 /usr/local/bin/adbcat << EOT > /usr/local/bin/fastboot
    /usr/local/platform-tools/fastboot “$@”
    chmod 755 /usr/local/bin/fastboot
  • Download and install a UDEV rules file which identifies most if not all known Android devices and makes udev create device-files for them when the Android connects (strictly speaking, his is only relevant for remote debugging over USB… if you connect over wireless then device-files are not used). The file is community-maintained and part of a git repository for Ubuntu. After installing the UDEV rules, you need to reload udevd’s cached rules:
    wget -O /etc/udev/rules.d/51-android.rules
    chmod 644 /etc/udev/rules.d/51-android.rules
    /etc/rc.d/rc.udev reload

You’re all set now to connect to your Android device and interact with it. Thanks to the UDEV rules you’ve installed, you won’t have to be root to use ‘adb‘ and ‘fastboot‘ as long as your user account is a member of the “plugdev” group.

Tip: if you have a USB stick with the Slackware Live Edition, you can repeat the above with one change: instead of installing the debugger tools to /usr/local , the wrapper scripts to /usr/local/bin and the UDEV rules to /etc/udev/rules.d/ you (i.e. with “/” as the root) you install the lot to the /rootcopy/ directory of the USB stick, keeping the same directory structure and thus ending with /rootcopy/usr/local/bin/ and /rootcopy/etc/udev/rules.d/ . That will make your Slackware Live a powerful tool to manage Android devices.

Enable remote adb debugging

Android devices do not allow remote debugging by default and it is not immediately visible how to enable this capability. You will have to enable developer options on your Android. This takes the form of a secret ritual… and enables several new menus and options in your Settings.
First, generic Android instructions:

  • Open the “Settings” on your Android.
  • Scroll down to “About phone” (or “About Tablet”)
  • Find the entry which shows the “Build number”.
  • Repeatedly, tap the “Build number” line. After a couple of taps, a popup message tells you that you are an “X amount” of clicks away from becoming a Developer. As you keep tapping the number “X” will decrease to zero which completes the process.

Android Wear instructions:

  • Swipe down from the top of the screen.
  • Tap the “Settings” icon (gear icon).
  • Scroll down and tap “System”.
  • Scroll down and tap “About”.
  • Then scroll down and tap “Build Number” repeatedly.
  • Keep tapping until you get the message “You’re now a developer”.
  • Swipe to the right twice.
  • Scroll down and you’ll see new “Developer Options”.

In the new Developer Options menu you will find “ADB debugging” which you have to enable. For the Android Wear device, it makes sense to also enable “Debug over Wi-Fi” since at least my watch does not have a micro-USB connector.

If you look at the screenshot you’ll notice that the watch’s IP address is shown in the “Debug over Wi-Fi” section. We will need that IP address soon.

Connect to Android using adb

Connecting to a device which is connected through USB is simple. You request an overview of detected devices and if your device (phone or tablet) is detected, it will be listed and you can control it remotely:

adb devices

With a smartwatch, you first need to connect to its IP addres over the port number shown in the screenshot above:

adb connect

You do not need to be root to do this.
Any of the above two commands will start the adb daemon if it is not yet running. In case of connecting to a smartwatch, you will have to accept the incoming connection on the watch-side.

You can now start sending commands to be executed on the Android device. For phones, this is usually part of the rooting process followed by installing a custom ROM. That is beyond the scope of this article.

Pair your watch with a new phone

Now that the pre-requisites have been taken care of, let’s go through the steps to pair the Android Wear watch with a new phone. You do not need to be root to do this.

  • Disable Bluetooth on your old (if you have it still) and new phone.
  • Open an adb shell – basically a remote shell on the watch. You’ll notice the prompt (sawshark:/ $), sawshark is the Android codename for my Huawei Watch 2 Sport:
    adb shell
    sawshark:/ $
  • Execute the command that instructs Package Manager to clear stored data & cache (“pm clear”) of the Google Play Services (“”) and reboot the watch:
    pm clear && reboot

    This clears the pairing information including the keys to communicate securely with the paired phone, but does not trigger a factory reset which happens if you un-pair the phone via the watch settings menu.

  • Make sure you have installed the Android Wear app on your new phone but do not enable Bluetooth on the phone yet.
  • From the computer, connect to your watch again over Wi-Fi (accept the incoming connection request on your watch) using the “adb connect” command shown earlier. Start a new “adb shell”.
  • At the remote prompt enter the command to make the watch visible for Bluetooth discovery by telling Activity Manager to start an Activity specified by the intent “android.bluetooth.adapter.action.REQUEST_DISCOVERABLE” (and make sure you accept the ‘OK’ prompt on your watch):
    am start -a android.bluetooth.adapter.action.REQUEST_DISCOVERABLE
  • Now is the time to start Android Wear on your new phone, allow the app to turn on Bluetooth and start scanning for a Android Wear watch to pair with. When the phone finds your watch, an accept request will show on both. Once you have accepted both prompts, new pairing keys will be generated and stored in the watch’s Google Play Services app. You can synchronize the watch data to your phone and manage it from there.

Good luck! Eric

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