Linux Sound Help And Tricks

Getting a computer running Linux to make the sounds you want it to make can be a real challenge. The problem seems inherent in the solution - open source software (and the Unix philosophy) tends to fragment into very specific functional components. So a low level hardware driver related function is cleanly separated from a user level function. With Linux sound there are actually many layers and some projects encompass several layers. This creates some choice, flexibility, developer modularity, and, well, confusion.

Here’s a good article on the complex organization of Linux sound.

Here is an excellent guide to troubleshooting Linux sound problems. (Hint, MM is mute and OO is not mute in alsamixer. Sounds obvious, but when everything goes mute suddenly for no discernible reason there can be uncertainty.)

The general technique for making mp3s from raw wav files is something like this:

One thing I sometimes want to do but often find quite strangely difficult is to simply record things. Sometimes there is a generic "sound recorder" utility ( "gnome-sound-recorder"?) present on some distros. If that works great. For quick command line work also think about rec from the sox package. I think the serious tools are Ardour and Audacity. Ardour seems great but I couldn’t get it to work.

I used Audacity to record all of my videos.

That leaves Audacity which also seems great. The classic problem is that there is OSS/ALSA/JACK/PA all fighting and emulating each other - a big mess. With Audacity you can go to Edit-→Devices and skip a lot of that.

One of the things I need to do is to record from my Boss GT10 Guitar Effects Processor. This capital piece of hardware has a USB port on it which presents itself to computers as would a USB sound device. So in Audacity I can just set the Edit-→Devices-→Recording-→Device to "GT-10: USB Audio (hw:1,0)". I can leave the Edit-→Devices-→Playback at "Default" and recording will be done from the guitar and playback will go to the headphones on the computer. What’s annoying and seems intractable is that I can’t monitor the guitar input in the headphones. If I figure that out I’ll note it but right now I can just click the "Recording Level" meter to monitor it graphically - but it’s not audible until recorded. The answer is to use headphones on the GT-10. This is ok since it will remove latency in the bargain. Then use headphones on the computer to verify the recording is ok and to edit it.

Imagine that you have two computers that can ping each other on some network (could be the wild internet, could be a home LAN). There is the computer you’re sitting at which we’ll call "home" and the one that’s not near you which we’ll call "away". Now imagine that you want to listen to what’s going on around "away" while sitting at "home".

Start by telling the "home" computer that you want it to wait for a network connection (which presumably will be from "away") and when it makes the connection to pipe it to something that will send the data to the speakers for you to hear. You need to choose a port, here I’ll use 7777. You need sudo to use ports lower than 1000.

That should appear to do pretty much nothing, but it’s actually waiting for a network connection from somewhere.

Next we log in to "away" and fire up the microphone and send it to home (substitute the name of your host in the home position).

The 7777 is the port number and can be whatever you like that is not being used or blocked by a firewall. The -b is for bits (per sample?). The -c1 is for mono channel. The -r 44100 is sampling rate. Another simpler setup might be -t raw -b8 -c1 -r8k - which are pretty modest and designed to keep things simple for efficient transfer of basic noises, especially talking. But maybe they don’t quite work. Moving a recording of a violin concerto over a network is a different problem.

I found that the sound quality was kind of scratchy. I made that better by recording some very quiet time on "away" and using that to generate a noise reduction profile that could then be used to filter the recording process.

It can happen that everything works fine in controlled testing but while deploying in a live situation, nothing works. It used to be that if some user was logged in and playing sound (or had a Flash thing going on in a browser) that other logged in users couldn’t access the sound device. Apparently this was cured and was only a problem for people who had legacy accounts which were members of the audio group. You can cure it by making sure no one is in the audio group:

Counter intuitive, but seems to work. Fixing this will allow this remote listening strategy to work even if someone else is logged onto "away" and busy watching videos or something.

Ok, these audio files are not cool. They often don’t work with mp3 infrastructure. Here’s how to convert them to mp3s:

I don’t really know if there’s a serious performance hit, but the resulting file sizes from the set I did were smaller. Probably just saved some space without any noticable loss of quality.

Here is a nice description of all the overly complex details of ALSA. Things like what does hw:0,1 mean? (Answer "first device on the second soundcard/device".)

Also to get useful information, try aplay -l or aplay -L or cat /proc/asound/cards

I tend to have problems getting mpg123 to work since that’s the player I tend to use. The trick is to do something like this:

Which option argument should you use after the -o? Try the --list-modules option to see. In the same style look at --test-cpu but pretty much everyone should have fancy decoding tricks in their CPUs by now.

Note that mpg123 can easily run into problems, but they are generally solvable. I often use this now.

Problem with the wrong mic or speakers or they’re just not working? Check out pavucontrol.

If you have an application that looks for /dev/dsp and doesn’t find it, check out the man page for padsp, a wrapper that sets the the LD_PRELOAD variable to be /usr/lib/x86_64-linux-gnu/pulseaudio/libpulsedsp.so and things magically start working.

To rip audio CDs from the command line (sometimes the only way I can figure out how to play the music on them) use abcde. This program works very well but note that it defaults to ogg. If you need mp3 for some player’s compatibility, try the -o mp3 option. This may want eyed3 and lame as dependency packages. To clean up the abominable file names that come from CDDB, try apt install detox. Run with detox -vr dir_filled_w_mp3s. Also -n for detox dry-run if you want to preview. Don’t confuse the program abcde with the following.

ABC notation is a way to comprehensively express music in ASCII. For details, check out this simple ABC primer or the full standard.

Standard Notation in XML It seems this is being actively worked on. The most standard of open standards, good to finally see some modern attention to the problem.

Linux Multimedia Studio "LMMS is a free cross-platform software which allows you to produce music with your computer. This covers creating melodies and beats, synthesizing and mixing sounds and arranging samples. You can have fun with your MIDI keyboard and much more - all in a user-friendly and modern interface. Furthermore LMMS comes with many ready-to-use instrument and effect plugins, presets and samples."

ZynAddSubFX A software synthesizer that allows one to create custom wave forms. Nice Demo here.

ReZound This is a very awesome sound editor. You can do all kinds of crazy things with this and I can only understand 1% of it. I have used it to crop sound files that had undesirable stuff at the ends, and isolate certain sounds in a longer sound file. I also managed to get it to slow down a song by 50% while bringing up the pitch by an octave so that I could try to figure out some guitar part. Unfortunately that produced a seg fault on my system when I tried to do too much at once. Probably my machine doesn’t have enough memory for this kind of thing. Still a great piece of software. Note that the right mouse button is used to define the right selection boundary. This can be a bit non-standard.

playitslowly This is a simple but quite functional program that basically plays songs at different speed while adjusting the pitch to remain constant. I have used this and it works great (contrast to my experience with rezound). It is ideal for figuring out lyrics, guitar riffs, and drum parts, etc.

SooperLooper "SooperLooper is a live looping sampler capable of immediate loop recording, overdubbing, multiplying, reversing and more. It allows for multiple simultaneous multi-channel loops limited only by your computer’s available memory." Uses JACK.

Hydrogen Drum "Hydrogen is an advanced drum machine for GNU/Linux. It’s main goal is to bring professional yet simple and intuitive pattern-based drum programming."

Freewheeling "Freewheeling allows us to build repetitive grooves by sampling and directing loops…"

kluppe "kluppe is a loop-player and recorder, designed for live use."

qtractor "Qtractor is an Audio/MIDI multi-track sequencer application written in C++ with the Qt4 framework. Target platform is Linux, where the Jack Audio Connection Kit (JACK) for audio, and the Advanced Linux Sound Architecture (ALSA) for MIDI, are the main infrastructures to evolve as a fairly-featured Linux desktop audio workstation GUI, specially dedicated to the personal home-studio."

Ardour "Digital Audio Workstation. record mix edit collaborate."

BEAST Stands for "BEtter Audio SysTem" and it comes with BSE, "Better Audio Engine" but it’s really an audio synthesizer and multi-track editor.

GTick "Tick is a metronome application written for GNU/Linux and other UN*X-like operting systems supporting different meters (Even, 2/4, 3/4, 4/4 and more) and speeds ranging from 10 to 1000 bpm. It utilizes GTK+ and OSS (ALSA compatible). It is part of the GNU Project."

Rosegarden "Rosegarden is a well-rounded audio and MIDI sequencer, score editor, and general-purpose music composition and editing environment. Rosegarden is an easy-to-learn, attractive application that runs on Linux, ideal for composers, musicians, music students, and small studio or home recording environments."

AlgoScore "AlgoScore is a graphical environment for algorithmic composition, where music is constructed directly in an interactive graphical score. The result is output as audio (through CSound), arbitrary control data (through JACK ports) for control of other applications, MIDI through JACK or to file, or OpenSoundControl messages. The generated audio can be played back through JACK or exported to an audiofile."

And many more!